AudioFlinger.cpp revision e0fa467e1150c65a7b1b1ed904c579b40f97c9df
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#undef ADD_BATTERY_DATA 41 42#ifdef ADD_BATTERY_DATA 43#include <media/IMediaPlayerService.h> 44#include <media/IMediaDeathNotifier.h> 45#endif 46 47#include <private/media/AudioTrackShared.h> 48#include <private/media/AudioEffectShared.h> 49 50#include <system/audio.h> 51#include <hardware/audio.h> 52 53#include "AudioMixer.h" 54#include "AudioFlinger.h" 55#include "ServiceUtilities.h" 56 57#include <media/EffectsFactoryApi.h> 58#include <audio_effects/effect_visualizer.h> 59#include <audio_effects/effect_ns.h> 60#include <audio_effects/effect_aec.h> 61 62#include <audio_utils/primitives.h> 63 64#include <powermanager/PowerManager.h> 65 66// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72#include <common_time/cc_helper.h> 73#include <common_time/local_clock.h> 74 75#include "FastMixer.h" 76 77// NBAIO implementations 78#include "AudioStreamOutSink.h" 79#include "MonoPipe.h" 80#include "MonoPipeReader.h" 81#include "SourceAudioBufferProvider.h" 82 83#ifdef HAVE_REQUEST_PRIORITY 84#include "SchedulingPolicyService.h" 85#endif 86 87#ifdef SOAKER 88#include "Soaker.h" 89#endif 90 91// ---------------------------------------------------------------------------- 92 93// Note: the following macro is used for extremely verbose logging message. In 94// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 95// 0; but one side effect of this is to turn all LOGV's as well. Some messages 96// are so verbose that we want to suppress them even when we have ALOG_ASSERT 97// turned on. Do not uncomment the #def below unless you really know what you 98// are doing and want to see all of the extremely verbose messages. 99//#define VERY_VERY_VERBOSE_LOGGING 100#ifdef VERY_VERY_VERBOSE_LOGGING 101#define ALOGVV ALOGV 102#else 103#define ALOGVV(a...) do { } while(0) 104#endif 105 106namespace android { 107 108static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 109static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 110 111static const float MAX_GAIN = 4096.0f; 112static const uint32_t MAX_GAIN_INT = 0x1000; 113 114// retry counts for buffer fill timeout 115// 50 * ~20msecs = 1 second 116static const int8_t kMaxTrackRetries = 50; 117static const int8_t kMaxTrackStartupRetries = 50; 118// allow less retry attempts on direct output thread. 119// direct outputs can be a scarce resource in audio hardware and should 120// be released as quickly as possible. 121static const int8_t kMaxTrackRetriesDirect = 2; 122 123static const int kDumpLockRetries = 50; 124static const int kDumpLockSleepUs = 20000; 125 126// don't warn about blocked writes or record buffer overflows more often than this 127static const nsecs_t kWarningThrottleNs = seconds(5); 128 129// RecordThread loop sleep time upon application overrun or audio HAL read error 130static const int kRecordThreadSleepUs = 5000; 131 132// maximum time to wait for setParameters to complete 133static const nsecs_t kSetParametersTimeoutNs = seconds(2); 134 135// minimum sleep time for the mixer thread loop when tracks are active but in underrun 136static const uint32_t kMinThreadSleepTimeUs = 5000; 137// maximum divider applied to the active sleep time in the mixer thread loop 138static const uint32_t kMaxThreadSleepTimeShift = 2; 139 140// minimum normal mix buffer size, expressed in milliseconds rather than frames 141static const uint32_t kMinNormalMixBufferSizeMs = 20; 142 143nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 161{ 162 const hw_module_t *mod; 163 int rc; 164 165 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 166 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 167 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 168 if (rc) { 169 goto out; 170 } 171 rc = audio_hw_device_open(mod, dev); 172 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 173 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 174 if (rc) { 175 goto out; 176 } 177 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 178 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 179 rc = BAD_VALUE; 180 goto out; 181 } 182 return 0; 183 184out: 185 *dev = NULL; 186 return rc; 187} 188 189// ---------------------------------------------------------------------------- 190 191AudioFlinger::AudioFlinger() 192 : BnAudioFlinger(), 193 mPrimaryHardwareDev(NULL), 194 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 195 mMasterVolume(1.0f), 196 mMasterVolumeSupportLvl(MVS_NONE), 197 mMasterMute(false), 198 mNextUniqueId(1), 199 mMode(AUDIO_MODE_INVALID), 200 mBtNrecIsOff(false) 201{ 202} 203 204void AudioFlinger::onFirstRef() 205{ 206 int rc = 0; 207 208 Mutex::Autolock _l(mLock); 209 210 /* TODO: move all this work into an Init() function */ 211 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 212 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 213 uint32_t int_val; 214 if (1 == sscanf(val_str, "%u", &int_val)) { 215 mStandbyTimeInNsecs = milliseconds(int_val); 216 ALOGI("Using %u mSec as standby time.", int_val); 217 } else { 218 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 219 ALOGI("Using default %u mSec as standby time.", 220 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 221 } 222 } 223 224 mMode = AUDIO_MODE_NORMAL; 225 mMasterVolumeSW = 1.0; 226 mMasterVolume = 1.0; 227 mHardwareStatus = AUDIO_HW_IDLE; 228} 229 230AudioFlinger::~AudioFlinger() 231{ 232 233 while (!mRecordThreads.isEmpty()) { 234 // closeInput() will remove first entry from mRecordThreads 235 closeInput(mRecordThreads.keyAt(0)); 236 } 237 while (!mPlaybackThreads.isEmpty()) { 238 // closeOutput() will remove first entry from mPlaybackThreads 239 closeOutput(mPlaybackThreads.keyAt(0)); 240 } 241 242 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 243 // no mHardwareLock needed, as there are no other references to this 244 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 245 delete mAudioHwDevs.valueAt(i); 246 } 247} 248 249static const char * const audio_interfaces[] = { 250 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 251 AUDIO_HARDWARE_MODULE_ID_A2DP, 252 AUDIO_HARDWARE_MODULE_ID_USB, 253}; 254#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 255 256audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 257{ 258 // if module is 0, the request comes from an old policy manager and we should load 259 // well known modules 260 if (module == 0) { 261 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 262 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 263 loadHwModule_l(audio_interfaces[i]); 264 } 265 } else { 266 // check a match for the requested module handle 267 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 268 if (audioHwdevice != NULL) { 269 return audioHwdevice->hwDevice(); 270 } 271 } 272 // then try to find a module supporting the requested device. 273 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 274 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 275 if ((dev->get_supported_devices(dev) & devices) == devices) 276 return dev; 277 } 278 279 return NULL; 280} 281 282status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 283{ 284 const size_t SIZE = 256; 285 char buffer[SIZE]; 286 String8 result; 287 288 result.append("Clients:\n"); 289 for (size_t i = 0; i < mClients.size(); ++i) { 290 sp<Client> client = mClients.valueAt(i).promote(); 291 if (client != 0) { 292 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 293 result.append(buffer); 294 } 295 } 296 297 result.append("Global session refs:\n"); 298 result.append(" session pid count\n"); 299 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 300 AudioSessionRef *r = mAudioSessionRefs[i]; 301 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 302 result.append(buffer); 303 } 304 write(fd, result.string(), result.size()); 305 return NO_ERROR; 306} 307 308 309status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 310{ 311 const size_t SIZE = 256; 312 char buffer[SIZE]; 313 String8 result; 314 hardware_call_state hardwareStatus = mHardwareStatus; 315 316 snprintf(buffer, SIZE, "Hardware status: %d\n" 317 "Standby Time mSec: %u\n", 318 hardwareStatus, 319 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 320 result.append(buffer); 321 write(fd, result.string(), result.size()); 322 return NO_ERROR; 323} 324 325status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 326{ 327 const size_t SIZE = 256; 328 char buffer[SIZE]; 329 String8 result; 330 snprintf(buffer, SIZE, "Permission Denial: " 331 "can't dump AudioFlinger from pid=%d, uid=%d\n", 332 IPCThreadState::self()->getCallingPid(), 333 IPCThreadState::self()->getCallingUid()); 334 result.append(buffer); 335 write(fd, result.string(), result.size()); 336 return NO_ERROR; 337} 338 339static bool tryLock(Mutex& mutex) 340{ 341 bool locked = false; 342 for (int i = 0; i < kDumpLockRetries; ++i) { 343 if (mutex.tryLock() == NO_ERROR) { 344 locked = true; 345 break; 346 } 347 usleep(kDumpLockSleepUs); 348 } 349 return locked; 350} 351 352status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 353{ 354 if (!dumpAllowed()) { 355 dumpPermissionDenial(fd, args); 356 } else { 357 // get state of hardware lock 358 bool hardwareLocked = tryLock(mHardwareLock); 359 if (!hardwareLocked) { 360 String8 result(kHardwareLockedString); 361 write(fd, result.string(), result.size()); 362 } else { 363 mHardwareLock.unlock(); 364 } 365 366 bool locked = tryLock(mLock); 367 368 // failed to lock - AudioFlinger is probably deadlocked 369 if (!locked) { 370 String8 result(kDeadlockedString); 371 write(fd, result.string(), result.size()); 372 } 373 374 dumpClients(fd, args); 375 dumpInternals(fd, args); 376 377 // dump playback threads 378 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 379 mPlaybackThreads.valueAt(i)->dump(fd, args); 380 } 381 382 // dump record threads 383 for (size_t i = 0; i < mRecordThreads.size(); i++) { 384 mRecordThreads.valueAt(i)->dump(fd, args); 385 } 386 387 // dump all hardware devs 388 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 389 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 390 dev->dump(dev, fd); 391 } 392 if (locked) mLock.unlock(); 393 } 394 return NO_ERROR; 395} 396 397sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 398{ 399 // If pid is already in the mClients wp<> map, then use that entry 400 // (for which promote() is always != 0), otherwise create a new entry and Client. 401 sp<Client> client = mClients.valueFor(pid).promote(); 402 if (client == 0) { 403 client = new Client(this, pid); 404 mClients.add(pid, client); 405 } 406 407 return client; 408} 409 410// IAudioFlinger interface 411 412 413sp<IAudioTrack> AudioFlinger::createTrack( 414 pid_t pid, 415 audio_stream_type_t streamType, 416 uint32_t sampleRate, 417 audio_format_t format, 418 uint32_t channelMask, 419 int frameCount, 420 IAudioFlinger::track_flags_t flags, 421 const sp<IMemory>& sharedBuffer, 422 audio_io_handle_t output, 423 pid_t tid, 424 int *sessionId, 425 status_t *status) 426{ 427 sp<PlaybackThread::Track> track; 428 sp<TrackHandle> trackHandle; 429 sp<Client> client; 430 status_t lStatus; 431 int lSessionId; 432 433 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 434 // but if someone uses binder directly they could bypass that and cause us to crash 435 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 436 ALOGE("createTrack() invalid stream type %d", streamType); 437 lStatus = BAD_VALUE; 438 goto Exit; 439 } 440 441 { 442 Mutex::Autolock _l(mLock); 443 PlaybackThread *thread = checkPlaybackThread_l(output); 444 PlaybackThread *effectThread = NULL; 445 if (thread == NULL) { 446 ALOGE("unknown output thread"); 447 lStatus = BAD_VALUE; 448 goto Exit; 449 } 450 451 client = registerPid_l(pid); 452 453 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 454 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 455 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 456 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 457 if (mPlaybackThreads.keyAt(i) != output) { 458 // prevent same audio session on different output threads 459 uint32_t sessions = t->hasAudioSession(*sessionId); 460 if (sessions & PlaybackThread::TRACK_SESSION) { 461 ALOGE("createTrack() session ID %d already in use", *sessionId); 462 lStatus = BAD_VALUE; 463 goto Exit; 464 } 465 // check if an effect with same session ID is waiting for a track to be created 466 if (sessions & PlaybackThread::EFFECT_SESSION) { 467 effectThread = t.get(); 468 } 469 } 470 } 471 lSessionId = *sessionId; 472 } else { 473 // if no audio session id is provided, create one here 474 lSessionId = nextUniqueId(); 475 if (sessionId != NULL) { 476 *sessionId = lSessionId; 477 } 478 } 479 ALOGV("createTrack() lSessionId: %d", lSessionId); 480 481 track = thread->createTrack_l(client, streamType, sampleRate, format, 482 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 483 484 // move effect chain to this output thread if an effect on same session was waiting 485 // for a track to be created 486 if (lStatus == NO_ERROR && effectThread != NULL) { 487 Mutex::Autolock _dl(thread->mLock); 488 Mutex::Autolock _sl(effectThread->mLock); 489 moveEffectChain_l(lSessionId, effectThread, thread, true); 490 } 491 492 // Look for sync events awaiting for a session to be used. 493 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 494 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 495 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 496 track->setSyncEvent(mPendingSyncEvents[i]); 497 mPendingSyncEvents.removeAt(i); 498 i--; 499 } 500 } 501 } 502 } 503 if (lStatus == NO_ERROR) { 504 trackHandle = new TrackHandle(track); 505 } else { 506 // remove local strong reference to Client before deleting the Track so that the Client 507 // destructor is called by the TrackBase destructor with mLock held 508 client.clear(); 509 track.clear(); 510 } 511 512Exit: 513 if (status != NULL) { 514 *status = lStatus; 515 } 516 return trackHandle; 517} 518 519uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 520{ 521 Mutex::Autolock _l(mLock); 522 PlaybackThread *thread = checkPlaybackThread_l(output); 523 if (thread == NULL) { 524 ALOGW("sampleRate() unknown thread %d", output); 525 return 0; 526 } 527 return thread->sampleRate(); 528} 529 530int AudioFlinger::channelCount(audio_io_handle_t output) const 531{ 532 Mutex::Autolock _l(mLock); 533 PlaybackThread *thread = checkPlaybackThread_l(output); 534 if (thread == NULL) { 535 ALOGW("channelCount() unknown thread %d", output); 536 return 0; 537 } 538 return thread->channelCount(); 539} 540 541audio_format_t AudioFlinger::format(audio_io_handle_t output) const 542{ 543 Mutex::Autolock _l(mLock); 544 PlaybackThread *thread = checkPlaybackThread_l(output); 545 if (thread == NULL) { 546 ALOGW("format() unknown thread %d", output); 547 return AUDIO_FORMAT_INVALID; 548 } 549 return thread->format(); 550} 551 552size_t AudioFlinger::frameCount(audio_io_handle_t output) const 553{ 554 Mutex::Autolock _l(mLock); 555 PlaybackThread *thread = checkPlaybackThread_l(output); 556 if (thread == NULL) { 557 ALOGW("frameCount() unknown thread %d", output); 558 return 0; 559 } 560 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 561 // should examine all callers and fix them to handle smaller counts 562 return thread->frameCount(); 563} 564 565uint32_t AudioFlinger::latency(audio_io_handle_t output) const 566{ 567 Mutex::Autolock _l(mLock); 568 PlaybackThread *thread = checkPlaybackThread_l(output); 569 if (thread == NULL) { 570 ALOGW("latency() unknown thread %d", output); 571 return 0; 572 } 573 return thread->latency(); 574} 575 576status_t AudioFlinger::setMasterVolume(float value) 577{ 578 status_t ret = initCheck(); 579 if (ret != NO_ERROR) { 580 return ret; 581 } 582 583 // check calling permissions 584 if (!settingsAllowed()) { 585 return PERMISSION_DENIED; 586 } 587 588 float swmv = value; 589 590 Mutex::Autolock _l(mLock); 591 592 // when hw supports master volume, don't scale in sw mixer 593 if (MVS_NONE != mMasterVolumeSupportLvl) { 594 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 595 AutoMutex lock(mHardwareLock); 596 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 597 598 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 599 if (NULL != dev->set_master_volume) { 600 dev->set_master_volume(dev, value); 601 } 602 mHardwareStatus = AUDIO_HW_IDLE; 603 } 604 605 swmv = 1.0; 606 } 607 608 mMasterVolume = value; 609 mMasterVolumeSW = swmv; 610 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 611 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 612 613 return NO_ERROR; 614} 615 616status_t AudioFlinger::setMode(audio_mode_t mode) 617{ 618 status_t ret = initCheck(); 619 if (ret != NO_ERROR) { 620 return ret; 621 } 622 623 // check calling permissions 624 if (!settingsAllowed()) { 625 return PERMISSION_DENIED; 626 } 627 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 628 ALOGW("Illegal value: setMode(%d)", mode); 629 return BAD_VALUE; 630 } 631 632 { // scope for the lock 633 AutoMutex lock(mHardwareLock); 634 mHardwareStatus = AUDIO_HW_SET_MODE; 635 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 636 mHardwareStatus = AUDIO_HW_IDLE; 637 } 638 639 if (NO_ERROR == ret) { 640 Mutex::Autolock _l(mLock); 641 mMode = mode; 642 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 643 mPlaybackThreads.valueAt(i)->setMode(mode); 644 } 645 646 return ret; 647} 648 649status_t AudioFlinger::setMicMute(bool state) 650{ 651 status_t ret = initCheck(); 652 if (ret != NO_ERROR) { 653 return ret; 654 } 655 656 // check calling permissions 657 if (!settingsAllowed()) { 658 return PERMISSION_DENIED; 659 } 660 661 AutoMutex lock(mHardwareLock); 662 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 663 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 664 mHardwareStatus = AUDIO_HW_IDLE; 665 return ret; 666} 667 668bool AudioFlinger::getMicMute() const 669{ 670 status_t ret = initCheck(); 671 if (ret != NO_ERROR) { 672 return false; 673 } 674 675 bool state = AUDIO_MODE_INVALID; 676 AutoMutex lock(mHardwareLock); 677 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 678 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 679 mHardwareStatus = AUDIO_HW_IDLE; 680 return state; 681} 682 683status_t AudioFlinger::setMasterMute(bool muted) 684{ 685 // check calling permissions 686 if (!settingsAllowed()) { 687 return PERMISSION_DENIED; 688 } 689 690 Mutex::Autolock _l(mLock); 691 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 692 mMasterMute = muted; 693 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 694 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 695 696 return NO_ERROR; 697} 698 699float AudioFlinger::masterVolume() const 700{ 701 Mutex::Autolock _l(mLock); 702 return masterVolume_l(); 703} 704 705float AudioFlinger::masterVolumeSW() const 706{ 707 Mutex::Autolock _l(mLock); 708 return masterVolumeSW_l(); 709} 710 711bool AudioFlinger::masterMute() const 712{ 713 Mutex::Autolock _l(mLock); 714 return masterMute_l(); 715} 716 717float AudioFlinger::masterVolume_l() const 718{ 719 if (MVS_FULL == mMasterVolumeSupportLvl) { 720 float ret_val; 721 AutoMutex lock(mHardwareLock); 722 723 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 724 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 725 (NULL != mPrimaryHardwareDev->get_master_volume), 726 "can't get master volume"); 727 728 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 729 mHardwareStatus = AUDIO_HW_IDLE; 730 return ret_val; 731 } 732 733 return mMasterVolume; 734} 735 736status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 737 audio_io_handle_t output) 738{ 739 // check calling permissions 740 if (!settingsAllowed()) { 741 return PERMISSION_DENIED; 742 } 743 744 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 745 ALOGE("setStreamVolume() invalid stream %d", stream); 746 return BAD_VALUE; 747 } 748 749 AutoMutex lock(mLock); 750 PlaybackThread *thread = NULL; 751 if (output) { 752 thread = checkPlaybackThread_l(output); 753 if (thread == NULL) { 754 return BAD_VALUE; 755 } 756 } 757 758 mStreamTypes[stream].volume = value; 759 760 if (thread == NULL) { 761 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 762 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 763 } 764 } else { 765 thread->setStreamVolume(stream, value); 766 } 767 768 return NO_ERROR; 769} 770 771status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 772{ 773 // check calling permissions 774 if (!settingsAllowed()) { 775 return PERMISSION_DENIED; 776 } 777 778 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 779 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 780 ALOGE("setStreamMute() invalid stream %d", stream); 781 return BAD_VALUE; 782 } 783 784 AutoMutex lock(mLock); 785 mStreamTypes[stream].mute = muted; 786 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 787 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 788 789 return NO_ERROR; 790} 791 792float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 793{ 794 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 795 return 0.0f; 796 } 797 798 AutoMutex lock(mLock); 799 float volume; 800 if (output) { 801 PlaybackThread *thread = checkPlaybackThread_l(output); 802 if (thread == NULL) { 803 return 0.0f; 804 } 805 volume = thread->streamVolume(stream); 806 } else { 807 volume = streamVolume_l(stream); 808 } 809 810 return volume; 811} 812 813bool AudioFlinger::streamMute(audio_stream_type_t stream) const 814{ 815 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 816 return true; 817 } 818 819 AutoMutex lock(mLock); 820 return streamMute_l(stream); 821} 822 823status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 824{ 825 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 826 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 827 // check calling permissions 828 if (!settingsAllowed()) { 829 return PERMISSION_DENIED; 830 } 831 832 // ioHandle == 0 means the parameters are global to the audio hardware interface 833 if (ioHandle == 0) { 834 Mutex::Autolock _l(mLock); 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.valueAt(i)->hwDevice(); 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 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 851 if (mBtNrecIsOff != btNrecIsOff) { 852 for (size_t i = 0; i < mRecordThreads.size(); i++) { 853 sp<RecordThread> thread = mRecordThreads.valueAt(i); 854 RecordThread::RecordTrack *track = thread->track(); 855 if (track != NULL) { 856 audio_devices_t device = (audio_devices_t)( 857 thread->device() & AUDIO_DEVICE_IN_ALL); 858 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 859 thread->setEffectSuspended(FX_IID_AEC, 860 suspend, 861 track->sessionId()); 862 thread->setEffectSuspended(FX_IID_NS, 863 suspend, 864 track->sessionId()); 865 } 866 } 867 mBtNrecIsOff = btNrecIsOff; 868 } 869 } 870 return final_result; 871 } 872 873 // hold a strong ref on thread in case closeOutput() or closeInput() is called 874 // and the thread is exited once the lock is released 875 sp<ThreadBase> thread; 876 { 877 Mutex::Autolock _l(mLock); 878 thread = checkPlaybackThread_l(ioHandle); 879 if (thread == NULL) { 880 thread = checkRecordThread_l(ioHandle); 881 } else if (thread == primaryPlaybackThread_l()) { 882 // indicate output device change to all input threads for pre processing 883 AudioParameter param = AudioParameter(keyValuePairs); 884 int value; 885 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 886 (value != 0)) { 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 Mutex::Autolock _l(mLock); 905 906 if (ioHandle == 0) { 907 String8 out_s8; 908 909 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 910 char *s; 911 { 912 AutoMutex lock(mHardwareLock); 913 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 914 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 915 s = dev->get_parameters(dev, keys.string()); 916 mHardwareStatus = AUDIO_HW_IDLE; 917 } 918 out_s8 += String8(s ? s : ""); 919 free(s); 920 } 921 return out_s8; 922 } 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 struct audio_config config = { 945 sample_rate: sampleRate, 946 channel_mask: audio_channel_in_mask_from_count(channelCount), 947 format: format, 948 }; 949 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 950 mHardwareStatus = AUDIO_HW_IDLE; 951 return size; 952} 953 954unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 955{ 956 if (ioHandle == 0) { 957 return 0; 958 } 959 960 Mutex::Autolock _l(mLock); 961 962 RecordThread *recordThread = checkRecordThread_l(ioHandle); 963 if (recordThread != NULL) { 964 return recordThread->getInputFramesLost(); 965 } 966 return 0; 967} 968 969status_t AudioFlinger::setVoiceVolume(float value) 970{ 971 status_t ret = initCheck(); 972 if (ret != NO_ERROR) { 973 return ret; 974 } 975 976 // check calling permissions 977 if (!settingsAllowed()) { 978 return PERMISSION_DENIED; 979 } 980 981 AutoMutex lock(mHardwareLock); 982 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 983 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 984 mHardwareStatus = AUDIO_HW_IDLE; 985 986 return ret; 987} 988 989status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 990 audio_io_handle_t output) const 991{ 992 status_t status; 993 994 Mutex::Autolock _l(mLock); 995 996 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 997 if (playbackThread != NULL) { 998 return playbackThread->getRenderPosition(halFrames, dspFrames); 999 } 1000 1001 return BAD_VALUE; 1002} 1003 1004void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1005{ 1006 1007 Mutex::Autolock _l(mLock); 1008 1009 pid_t pid = IPCThreadState::self()->getCallingPid(); 1010 if (mNotificationClients.indexOfKey(pid) < 0) { 1011 sp<NotificationClient> notificationClient = new NotificationClient(this, 1012 client, 1013 pid); 1014 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1015 1016 mNotificationClients.add(pid, notificationClient); 1017 1018 sp<IBinder> binder = client->asBinder(); 1019 binder->linkToDeath(notificationClient); 1020 1021 // the config change is always sent from playback or record threads to avoid deadlock 1022 // with AudioSystem::gLock 1023 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1024 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1025 } 1026 1027 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1028 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1029 } 1030 } 1031} 1032 1033void AudioFlinger::removeNotificationClient(pid_t pid) 1034{ 1035 Mutex::Autolock _l(mLock); 1036 1037 mNotificationClients.removeItem(pid); 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->mPid, i); 1045 if (ref->mPid == pid) { 1046 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 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, const 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), mNormalFrameCount(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, "io handle: %d\n", mId); 1196 result.append(buffer); 1197 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1198 result.append(buffer); 1199 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1200 result.append(buffer); 1201 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1202 result.append(buffer); 1203 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1204 result.append(buffer); 1205 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1206 result.append(buffer); 1207 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1208 result.append(buffer); 1209 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1210 result.append(buffer); 1211 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1212 result.append(buffer); 1213 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1214 result.append(buffer); 1215 1216 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1217 result.append(buffer); 1218 result.append(" Index Command"); 1219 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1220 snprintf(buffer, SIZE, "\n %02d ", i); 1221 result.append(buffer); 1222 result.append(mNewParameters[i]); 1223 } 1224 1225 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1226 result.append(buffer); 1227 snprintf(buffer, SIZE, " Index event param\n"); 1228 result.append(buffer); 1229 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1230 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1231 result.append(buffer); 1232 } 1233 result.append("\n"); 1234 1235 write(fd, result.string(), result.size()); 1236 1237 if (locked) { 1238 mLock.unlock(); 1239 } 1240 return NO_ERROR; 1241} 1242 1243status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1244{ 1245 const size_t SIZE = 256; 1246 char buffer[SIZE]; 1247 String8 result; 1248 1249 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1250 write(fd, buffer, strlen(buffer)); 1251 1252 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1253 sp<EffectChain> chain = mEffectChains[i]; 1254 if (chain != 0) { 1255 chain->dump(fd, args); 1256 } 1257 } 1258 return NO_ERROR; 1259} 1260 1261void AudioFlinger::ThreadBase::acquireWakeLock() 1262{ 1263 Mutex::Autolock _l(mLock); 1264 acquireWakeLock_l(); 1265} 1266 1267void AudioFlinger::ThreadBase::acquireWakeLock_l() 1268{ 1269 if (mPowerManager == 0) { 1270 // use checkService() to avoid blocking if power service is not up yet 1271 sp<IBinder> binder = 1272 defaultServiceManager()->checkService(String16("power")); 1273 if (binder == 0) { 1274 ALOGW("Thread %s cannot connect to the power manager service", mName); 1275 } else { 1276 mPowerManager = interface_cast<IPowerManager>(binder); 1277 binder->linkToDeath(mDeathRecipient); 1278 } 1279 } 1280 if (mPowerManager != 0) { 1281 sp<IBinder> binder = new BBinder(); 1282 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1283 binder, 1284 String16(mName)); 1285 if (status == NO_ERROR) { 1286 mWakeLockToken = binder; 1287 } 1288 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1289 } 1290} 1291 1292void AudioFlinger::ThreadBase::releaseWakeLock() 1293{ 1294 Mutex::Autolock _l(mLock); 1295 releaseWakeLock_l(); 1296} 1297 1298void AudioFlinger::ThreadBase::releaseWakeLock_l() 1299{ 1300 if (mWakeLockToken != 0) { 1301 ALOGV("releaseWakeLock_l() %s", mName); 1302 if (mPowerManager != 0) { 1303 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1304 } 1305 mWakeLockToken.clear(); 1306 } 1307} 1308 1309void AudioFlinger::ThreadBase::clearPowerManager() 1310{ 1311 Mutex::Autolock _l(mLock); 1312 releaseWakeLock_l(); 1313 mPowerManager.clear(); 1314} 1315 1316void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1317{ 1318 sp<ThreadBase> thread = mThread.promote(); 1319 if (thread != 0) { 1320 thread->clearPowerManager(); 1321 } 1322 ALOGW("power manager service died !!!"); 1323} 1324 1325void AudioFlinger::ThreadBase::setEffectSuspended( 1326 const effect_uuid_t *type, bool suspend, int sessionId) 1327{ 1328 Mutex::Autolock _l(mLock); 1329 setEffectSuspended_l(type, suspend, sessionId); 1330} 1331 1332void AudioFlinger::ThreadBase::setEffectSuspended_l( 1333 const effect_uuid_t *type, bool suspend, int sessionId) 1334{ 1335 sp<EffectChain> chain = getEffectChain_l(sessionId); 1336 if (chain != 0) { 1337 if (type != NULL) { 1338 chain->setEffectSuspended_l(type, suspend); 1339 } else { 1340 chain->setEffectSuspendedAll_l(suspend); 1341 } 1342 } 1343 1344 updateSuspendedSessions_l(type, suspend, sessionId); 1345} 1346 1347void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1348{ 1349 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1350 if (index < 0) { 1351 return; 1352 } 1353 1354 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1355 mSuspendedSessions.editValueAt(index); 1356 1357 for (size_t i = 0; i < sessionEffects.size(); i++) { 1358 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1359 for (int j = 0; j < desc->mRefCount; j++) { 1360 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1361 chain->setEffectSuspendedAll_l(true); 1362 } else { 1363 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1364 desc->mType.timeLow); 1365 chain->setEffectSuspended_l(&desc->mType, true); 1366 } 1367 } 1368 } 1369} 1370 1371void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1372 bool suspend, 1373 int sessionId) 1374{ 1375 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1376 1377 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1378 1379 if (suspend) { 1380 if (index >= 0) { 1381 sessionEffects = mSuspendedSessions.editValueAt(index); 1382 } else { 1383 mSuspendedSessions.add(sessionId, sessionEffects); 1384 } 1385 } else { 1386 if (index < 0) { 1387 return; 1388 } 1389 sessionEffects = mSuspendedSessions.editValueAt(index); 1390 } 1391 1392 1393 int key = EffectChain::kKeyForSuspendAll; 1394 if (type != NULL) { 1395 key = type->timeLow; 1396 } 1397 index = sessionEffects.indexOfKey(key); 1398 1399 sp<SuspendedSessionDesc> desc; 1400 if (suspend) { 1401 if (index >= 0) { 1402 desc = sessionEffects.valueAt(index); 1403 } else { 1404 desc = new SuspendedSessionDesc(); 1405 if (type != NULL) { 1406 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1407 } 1408 sessionEffects.add(key, desc); 1409 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1410 } 1411 desc->mRefCount++; 1412 } else { 1413 if (index < 0) { 1414 return; 1415 } 1416 desc = sessionEffects.valueAt(index); 1417 if (--desc->mRefCount == 0) { 1418 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1419 sessionEffects.removeItemsAt(index); 1420 if (sessionEffects.isEmpty()) { 1421 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1422 sessionId); 1423 mSuspendedSessions.removeItem(sessionId); 1424 } 1425 } 1426 } 1427 if (!sessionEffects.isEmpty()) { 1428 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1429 } 1430} 1431 1432void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1433 bool enabled, 1434 int sessionId) 1435{ 1436 Mutex::Autolock _l(mLock); 1437 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1438} 1439 1440void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1441 bool enabled, 1442 int sessionId) 1443{ 1444 if (mType != RECORD) { 1445 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1446 // another session. This gives the priority to well behaved effect control panels 1447 // and applications not using global effects. 1448 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1449 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1450 } 1451 } 1452 1453 sp<EffectChain> chain = getEffectChain_l(sessionId); 1454 if (chain != 0) { 1455 chain->checkSuspendOnEffectEnabled(effect, enabled); 1456 } 1457} 1458 1459// ---------------------------------------------------------------------------- 1460 1461AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1462 AudioStreamOut* output, 1463 audio_io_handle_t id, 1464 uint32_t device, 1465 type_t type) 1466 : ThreadBase(audioFlinger, id, device, type), 1467 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1468 // Assumes constructor is called by AudioFlinger with it's mLock held, 1469 // but it would be safer to explicitly pass initial masterMute as parameter 1470 mMasterMute(audioFlinger->masterMute_l()), 1471 // mStreamTypes[] initialized in constructor body 1472 mOutput(output), 1473 // Assumes constructor is called by AudioFlinger with it's mLock held, 1474 // but it would be safer to explicitly pass initial masterVolume as parameter 1475 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1476 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1477 mMixerStatus(MIXER_IDLE), 1478 mPrevMixerStatus(MIXER_IDLE), 1479 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1480 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1481 mFastTrackNewMask(0) 1482{ 1483#if !LOG_NDEBUG 1484 memset(mFastTrackNewArray, 0, sizeof(mFastTrackNewArray)); 1485#endif 1486 snprintf(mName, kNameLength, "AudioOut_%X", id); 1487 1488 readOutputParameters(); 1489 1490 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1491 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1492 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1493 stream = (audio_stream_type_t) (stream + 1)) { 1494 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1495 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1496 } 1497 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1498 // because mAudioFlinger doesn't have one to copy from 1499} 1500 1501AudioFlinger::PlaybackThread::~PlaybackThread() 1502{ 1503 delete [] mMixBuffer; 1504} 1505 1506status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1507{ 1508 dumpInternals(fd, args); 1509 dumpTracks(fd, args); 1510 dumpEffectChains(fd, args); 1511 return NO_ERROR; 1512} 1513 1514status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1515{ 1516 const size_t SIZE = 256; 1517 char buffer[SIZE]; 1518 String8 result; 1519 1520 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1521 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1522 const stream_type_t *st = &mStreamTypes[i]; 1523 if (i > 0) { 1524 result.appendFormat(", "); 1525 } 1526 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1527 if (st->mute) { 1528 result.append("M"); 1529 } 1530 } 1531 result.append("\n"); 1532 write(fd, result.string(), result.length()); 1533 result.clear(); 1534 1535 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1536 result.append(buffer); 1537 result.append(" Name Client Type Fmt Chn mask Session Frames S M F SRate L dB R dB " 1538 "Server User Main buf Aux Buf\n"); 1539 for (size_t i = 0; i < mTracks.size(); ++i) { 1540 sp<Track> track = mTracks[i]; 1541 if (track != 0) { 1542 track->dump(buffer, SIZE); 1543 result.append(buffer); 1544 } 1545 } 1546 1547 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1548 result.append(buffer); 1549 result.append(" Name Client Type Fmt Chn mask Session Frames S M F SRate L dB R dB " 1550 "Server User Main buf Aux Buf\n"); 1551 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1552 sp<Track> track = mActiveTracks[i].promote(); 1553 if (track != 0) { 1554 track->dump(buffer, SIZE); 1555 result.append(buffer); 1556 } 1557 } 1558 write(fd, result.string(), result.size()); 1559 return NO_ERROR; 1560} 1561 1562status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1563{ 1564 const size_t SIZE = 256; 1565 char buffer[SIZE]; 1566 String8 result; 1567 1568 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1569 result.append(buffer); 1570 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1571 result.append(buffer); 1572 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1573 result.append(buffer); 1574 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1575 result.append(buffer); 1576 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1577 result.append(buffer); 1578 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1579 result.append(buffer); 1580 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1581 result.append(buffer); 1582 write(fd, result.string(), result.size()); 1583 1584 dumpBase(fd, args); 1585 1586 return NO_ERROR; 1587} 1588 1589// Thread virtuals 1590status_t AudioFlinger::PlaybackThread::readyToRun() 1591{ 1592 status_t status = initCheck(); 1593 if (status == NO_ERROR) { 1594 ALOGI("AudioFlinger's thread %p ready to run", this); 1595 } else { 1596 ALOGE("No working audio driver found."); 1597 } 1598 return status; 1599} 1600 1601void AudioFlinger::PlaybackThread::onFirstRef() 1602{ 1603 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1604} 1605 1606// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1607sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1608 const sp<AudioFlinger::Client>& client, 1609 audio_stream_type_t streamType, 1610 uint32_t sampleRate, 1611 audio_format_t format, 1612 uint32_t channelMask, 1613 int frameCount, 1614 const sp<IMemory>& sharedBuffer, 1615 int sessionId, 1616 IAudioFlinger::track_flags_t flags, 1617 pid_t tid, 1618 status_t *status) 1619{ 1620 sp<Track> track; 1621 status_t lStatus; 1622 1623 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1624 1625 // client expresses a preference for FAST, but we get the final say 1626 if (flags & IAudioFlinger::TRACK_FAST) { 1627 if ( 1628 // not timed 1629 (!isTimed) && 1630 // either of these use cases: 1631 ( 1632 // use case 1: shared buffer with any frame count 1633 ( 1634 (sharedBuffer != 0) 1635 ) || 1636 // use case 2: callback handler and frame count is default or at least as large as HAL 1637 ( 1638 (tid != -1) && 1639 ((frameCount == 0) || 1640 (frameCount >= (int) mFrameCount)) // FIXME int cast is due to wrong parameter type 1641 ) 1642 ) && 1643 // PCM data 1644 audio_is_linear_pcm(format) && 1645 // mono or stereo 1646 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1647 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1648#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1649 // hardware sample rate 1650 (sampleRate == mSampleRate) && 1651#endif 1652 // normal mixer has an associated fast mixer 1653 hasFastMixer() && 1654 // there are sufficient fast track slots available 1655 (mFastTrackAvailMask != 0) 1656 // FIXME test that MixerThread for this fast track has a capable output HAL 1657 // FIXME add a permission test also? 1658 ) { 1659 ALOGI("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1660 frameCount, mFrameCount); 1661 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1662 if (frameCount == 0) { 1663 frameCount = mFrameCount; 1664 } 1665 } else { 1666 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1667 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1668 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1669 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1670 audio_is_linear_pcm(format), 1671 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1672 flags &= ~IAudioFlinger::TRACK_FAST; 1673 // For compatibility with AudioTrack calculation, buffer depth is forced 1674 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1675 // This is probably too conservative, but legacy application code may depend on it. 1676 // If you change this calculation, also review the start threshold which is related. 1677 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1678 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1679 if (minBufCount < 2) { 1680 minBufCount = 2; 1681 } 1682 int minFrameCount = mNormalFrameCount * minBufCount; 1683 if (frameCount < minFrameCount) { 1684 frameCount = minFrameCount; 1685 } 1686 } 1687 } 1688 1689 if (mType == DIRECT) { 1690 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1691 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1692 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1693 "for output %p with format %d", 1694 sampleRate, format, channelMask, mOutput, mFormat); 1695 lStatus = BAD_VALUE; 1696 goto Exit; 1697 } 1698 } 1699 } else { 1700 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1701 if (sampleRate > mSampleRate*2) { 1702 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1703 lStatus = BAD_VALUE; 1704 goto Exit; 1705 } 1706 } 1707 1708 lStatus = initCheck(); 1709 if (lStatus != NO_ERROR) { 1710 ALOGE("Audio driver not initialized."); 1711 goto Exit; 1712 } 1713 1714 { // scope for mLock 1715 Mutex::Autolock _l(mLock); 1716 1717 // all tracks in same audio session must share the same routing strategy otherwise 1718 // conflicts will happen when tracks are moved from one output to another by audio policy 1719 // manager 1720 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1721 for (size_t i = 0; i < mTracks.size(); ++i) { 1722 sp<Track> t = mTracks[i]; 1723 if (t != 0 && !t->isOutputTrack()) { 1724 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1725 if (sessionId == t->sessionId() && strategy != actual) { 1726 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1727 strategy, actual); 1728 lStatus = BAD_VALUE; 1729 goto Exit; 1730 } 1731 } 1732 } 1733 1734 if (!isTimed) { 1735 track = new Track(this, client, streamType, sampleRate, format, 1736 channelMask, frameCount, sharedBuffer, sessionId, flags); 1737 } else { 1738 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1739 channelMask, frameCount, sharedBuffer, sessionId); 1740 } 1741 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1742 lStatus = NO_MEMORY; 1743 goto Exit; 1744 } 1745 mTracks.add(track); 1746 1747 sp<EffectChain> chain = getEffectChain_l(sessionId); 1748 if (chain != 0) { 1749 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1750 track->setMainBuffer(chain->inBuffer()); 1751 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1752 chain->incTrackCnt(); 1753 } 1754 } 1755 1756#ifdef HAVE_REQUEST_PRIORITY 1757 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1758 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1759 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1760 // so ask activity manager to do this on our behalf 1761 int err = requestPriority(callingPid, tid, 1); 1762 if (err != 0) { 1763 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1764 1, callingPid, tid, err); 1765 } 1766 } 1767#endif 1768 1769 lStatus = NO_ERROR; 1770 1771Exit: 1772 if (status) { 1773 *status = lStatus; 1774 } 1775 return track; 1776} 1777 1778uint32_t AudioFlinger::PlaybackThread::latency() const 1779{ 1780 Mutex::Autolock _l(mLock); 1781 if (initCheck() == NO_ERROR) { 1782 return mOutput->stream->get_latency(mOutput->stream); 1783 } else { 1784 return 0; 1785 } 1786} 1787 1788void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1789{ 1790 Mutex::Autolock _l(mLock); 1791 mMasterVolume = value; 1792} 1793 1794void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1795{ 1796 Mutex::Autolock _l(mLock); 1797 setMasterMute_l(muted); 1798} 1799 1800void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1801{ 1802 Mutex::Autolock _l(mLock); 1803 mStreamTypes[stream].volume = value; 1804} 1805 1806void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1807{ 1808 Mutex::Autolock _l(mLock); 1809 mStreamTypes[stream].mute = muted; 1810} 1811 1812float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1813{ 1814 Mutex::Autolock _l(mLock); 1815 return mStreamTypes[stream].volume; 1816} 1817 1818// addTrack_l() must be called with ThreadBase::mLock held 1819status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1820{ 1821 status_t status = ALREADY_EXISTS; 1822 1823 // set retry count for buffer fill 1824 track->mRetryCount = kMaxTrackStartupRetries; 1825 if (mActiveTracks.indexOf(track) < 0) { 1826 // the track is newly added, make sure it fills up all its 1827 // buffers before playing. This is to ensure the client will 1828 // effectively get the latency it requested. 1829 track->mFillingUpStatus = Track::FS_FILLING; 1830 track->mResetDone = false; 1831 mActiveTracks.add(track); 1832 if (track->mainBuffer() != mMixBuffer) { 1833 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1834 if (chain != 0) { 1835 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1836 chain->incActiveTrackCnt(); 1837 } 1838 } 1839 1840 status = NO_ERROR; 1841 } 1842 1843 ALOGV("mWaitWorkCV.broadcast"); 1844 mWaitWorkCV.broadcast(); 1845 1846 return status; 1847} 1848 1849// destroyTrack_l() must be called with ThreadBase::mLock held 1850void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1851{ 1852 track->mState = TrackBase::TERMINATED; 1853 if (mActiveTracks.indexOf(track) < 0) { 1854 removeTrack_l(track); 1855 } 1856} 1857 1858void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1859{ 1860 mTracks.remove(track); 1861 deleteTrackName_l(track->name()); 1862 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1863 if (chain != 0) { 1864 chain->decTrackCnt(); 1865 } 1866} 1867 1868String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1869{ 1870 String8 out_s8 = String8(""); 1871 char *s; 1872 1873 Mutex::Autolock _l(mLock); 1874 if (initCheck() != NO_ERROR) { 1875 return out_s8; 1876 } 1877 1878 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1879 out_s8 = String8(s); 1880 free(s); 1881 return out_s8; 1882} 1883 1884// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1885void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1886 AudioSystem::OutputDescriptor desc; 1887 void *param2 = NULL; 1888 1889 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1890 1891 switch (event) { 1892 case AudioSystem::OUTPUT_OPENED: 1893 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1894 desc.channels = mChannelMask; 1895 desc.samplingRate = mSampleRate; 1896 desc.format = mFormat; 1897 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1898 desc.latency = latency(); 1899 param2 = &desc; 1900 break; 1901 1902 case AudioSystem::STREAM_CONFIG_CHANGED: 1903 param2 = ¶m; 1904 case AudioSystem::OUTPUT_CLOSED: 1905 default: 1906 break; 1907 } 1908 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1909} 1910 1911void AudioFlinger::PlaybackThread::readOutputParameters() 1912{ 1913 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1914 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1915 mChannelCount = (uint16_t)popcount(mChannelMask); 1916 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1917 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1918 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1919 if (mFrameCount & 15) { 1920 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1921 mFrameCount); 1922 } 1923 1924 // Calculate size of normal mix buffer 1925 if (mType == MIXER) { 1926 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1927 mNormalFrameCount = ((minNormalFrameCount + mFrameCount - 1) / mFrameCount) * mFrameCount; 1928 if (mNormalFrameCount & 15) { 1929 ALOGW("Normal mix buffer size is %u frames but AudioMixer requires multiples of 16 " 1930 "frames", mNormalFrameCount); 1931 } 1932 } else { 1933 mNormalFrameCount = mFrameCount; 1934 } 1935 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 1936 1937 // FIXME - Current mixer implementation only supports stereo output: Always 1938 // Allocate a stereo buffer even if HW output is mono. 1939 delete[] mMixBuffer; 1940 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 1941 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 1942 1943 // force reconfiguration of effect chains and engines to take new buffer size and audio 1944 // parameters into account 1945 // Note that mLock is not held when readOutputParameters() is called from the constructor 1946 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1947 // matter. 1948 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1949 Vector< sp<EffectChain> > effectChains = mEffectChains; 1950 for (size_t i = 0; i < effectChains.size(); i ++) { 1951 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1952 } 1953} 1954 1955status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1956{ 1957 if (halFrames == NULL || dspFrames == NULL) { 1958 return BAD_VALUE; 1959 } 1960 Mutex::Autolock _l(mLock); 1961 if (initCheck() != NO_ERROR) { 1962 return INVALID_OPERATION; 1963 } 1964 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1965 1966 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1967} 1968 1969uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1970{ 1971 Mutex::Autolock _l(mLock); 1972 uint32_t result = 0; 1973 if (getEffectChain_l(sessionId) != 0) { 1974 result = EFFECT_SESSION; 1975 } 1976 1977 for (size_t i = 0; i < mTracks.size(); ++i) { 1978 sp<Track> track = mTracks[i]; 1979 if (sessionId == track->sessionId() && 1980 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1981 result |= TRACK_SESSION; 1982 break; 1983 } 1984 } 1985 1986 return result; 1987} 1988 1989uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1990{ 1991 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1992 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1993 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1994 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1995 } 1996 for (size_t i = 0; i < mTracks.size(); i++) { 1997 sp<Track> track = mTracks[i]; 1998 if (sessionId == track->sessionId() && 1999 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2000 return AudioSystem::getStrategyForStream(track->streamType()); 2001 } 2002 } 2003 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2004} 2005 2006 2007AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2008{ 2009 Mutex::Autolock _l(mLock); 2010 return mOutput; 2011} 2012 2013AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2014{ 2015 Mutex::Autolock _l(mLock); 2016 AudioStreamOut *output = mOutput; 2017 mOutput = NULL; 2018 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2019 // must push a NULL and wait for ack 2020 mOutputSink.clear(); 2021 mPipeSink.clear(); 2022 mNormalSink.clear(); 2023 return output; 2024} 2025 2026// this method must always be called either with ThreadBase mLock held or inside the thread loop 2027audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2028{ 2029 if (mOutput == NULL) { 2030 return NULL; 2031 } 2032 return &mOutput->stream->common; 2033} 2034 2035uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2036{ 2037 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2038 // decoding and transfer time. So sleeping for half of the latency would likely cause 2039 // underruns 2040 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2041 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2042 } else { 2043 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2044 } 2045} 2046 2047status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2048{ 2049 if (!isValidSyncEvent(event)) { 2050 return BAD_VALUE; 2051 } 2052 2053 Mutex::Autolock _l(mLock); 2054 2055 for (size_t i = 0; i < mTracks.size(); ++i) { 2056 sp<Track> track = mTracks[i]; 2057 if (event->triggerSession() == track->sessionId()) { 2058 track->setSyncEvent(event); 2059 return NO_ERROR; 2060 } 2061 } 2062 2063 return NAME_NOT_FOUND; 2064} 2065 2066bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2067{ 2068 switch (event->type()) { 2069 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2070 return true; 2071 default: 2072 break; 2073 } 2074 return false; 2075} 2076 2077// ---------------------------------------------------------------------------- 2078 2079AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2080 audio_io_handle_t id, uint32_t device, type_t type) 2081 : PlaybackThread(audioFlinger, output, id, device, type), 2082 // mAudioMixer below 2083#ifdef SOAKER 2084 mSoaker(NULL), 2085#endif 2086 // mFastMixer below 2087 mFastMixerFutex(0) 2088 // mOutputSink below 2089 // mPipeSink below 2090 // mNormalSink below 2091{ 2092 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2093 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2094 "mFrameCount=%d, mNormalFrameCount=%d", 2095 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2096 mNormalFrameCount); 2097 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2098 2099 // FIXME - Current mixer implementation only supports stereo output 2100 if (mChannelCount == 1) { 2101 ALOGE("Invalid audio hardware channel count"); 2102 } 2103 2104 // create an NBAIO sink for the HAL output stream, and negotiate 2105 mOutputSink = new AudioStreamOutSink(output->stream); 2106 size_t numCounterOffers = 0; 2107 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2108 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2109 ALOG_ASSERT(index == 0); 2110 2111 // initialize fast mixer if needed 2112 if (mFrameCount < mNormalFrameCount) { 2113 2114 // create a MonoPipe to connect our submix to FastMixer 2115 NBAIO_Format format = mOutputSink->format(); 2116 // frame count will be rounded up to a power of 2, so this formula should work well 2117 MonoPipe *monoPipe = new MonoPipe((mNormalFrameCount * 3) / 2, format, 2118 true /*writeCanBlock*/); 2119 const NBAIO_Format offers[1] = {format}; 2120 size_t numCounterOffers = 0; 2121 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2122 ALOG_ASSERT(index == 0); 2123 mPipeSink = monoPipe; 2124 2125#ifdef SOAKER 2126 // create a soaker as workaround for governor issues 2127 mSoaker = new Soaker(); 2128 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2129 mSoaker->run("Soaker", PRIORITY_LOWEST); 2130#endif 2131 2132 // create fast mixer and configure it initially with just one fast track for our submix 2133 mFastMixer = new FastMixer(); 2134 FastMixerStateQueue *sq = mFastMixer->sq(); 2135 FastMixerState *state = sq->begin(); 2136 FastTrack *fastTrack = &state->mFastTracks[0]; 2137 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2138 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2139 fastTrack->mVolumeProvider = NULL; 2140 fastTrack->mGeneration++; 2141 state->mFastTracksGen++; 2142 state->mTrackMask = 1; 2143 // fast mixer will use the HAL output sink 2144 state->mOutputSink = mOutputSink.get(); 2145 state->mOutputSinkGen++; 2146 state->mFrameCount = mFrameCount; 2147 state->mCommand = FastMixerState::COLD_IDLE; 2148 // already done in constructor initialization list 2149 //mFastMixerFutex = 0; 2150 state->mColdFutexAddr = &mFastMixerFutex; 2151 state->mColdGen++; 2152 state->mDumpState = &mFastMixerDumpState; 2153 sq->end(); 2154 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2155 2156 // start the fast mixer 2157 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2158#ifdef HAVE_REQUEST_PRIORITY 2159 pid_t tid = mFastMixer->getTid(); 2160 int err = requestPriority(getpid_cached, tid, 2); 2161 if (err != 0) { 2162 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2163 2, getpid_cached, tid, err); 2164 } 2165#endif 2166 2167 } else { 2168 mFastMixer = NULL; 2169 } 2170 mNormalSink = mOutputSink; 2171} 2172 2173AudioFlinger::MixerThread::~MixerThread() 2174{ 2175 if (mFastMixer != NULL) { 2176 FastMixerStateQueue *sq = mFastMixer->sq(); 2177 FastMixerState *state = sq->begin(); 2178 if (state->mCommand == FastMixerState::COLD_IDLE) { 2179 int32_t old = android_atomic_inc(&mFastMixerFutex); 2180 if (old == -1) { 2181 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2182 } 2183 } 2184 state->mCommand = FastMixerState::EXIT; 2185 sq->end(); 2186 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2187 mFastMixer->join(); 2188 // Though the fast mixer thread has exited, it's state queue is still valid. 2189 // We'll use that extract the final state which contains one remaining fast track 2190 // corresponding to our sub-mix. 2191 state = sq->begin(); 2192 ALOG_ASSERT(state->mTrackMask == 1); 2193 FastTrack *fastTrack = &state->mFastTracks[0]; 2194 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2195 delete fastTrack->mBufferProvider; 2196 sq->end(false /*didModify*/); 2197 delete mFastMixer; 2198#ifdef SOAKER 2199 if (mSoaker != NULL) { 2200 mSoaker->requestExitAndWait(); 2201 } 2202 delete mSoaker; 2203#endif 2204 } 2205 delete mAudioMixer; 2206} 2207 2208class CpuStats { 2209public: 2210 CpuStats(); 2211 void sample(const String8 &title); 2212#ifdef DEBUG_CPU_USAGE 2213private: 2214 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2215 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2216 2217 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2218 2219 int mCpuNum; // thread's current CPU number 2220 int mCpukHz; // frequency of thread's current CPU in kHz 2221#endif 2222}; 2223 2224CpuStats::CpuStats() 2225#ifdef DEBUG_CPU_USAGE 2226 : mCpuNum(-1), mCpukHz(-1) 2227#endif 2228{ 2229} 2230 2231void CpuStats::sample(const String8 &title) { 2232#ifdef DEBUG_CPU_USAGE 2233 // get current thread's delta CPU time in wall clock ns 2234 double wcNs; 2235 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2236 2237 // record sample for wall clock statistics 2238 if (valid) { 2239 mWcStats.sample(wcNs); 2240 } 2241 2242 // get the current CPU number 2243 int cpuNum = sched_getcpu(); 2244 2245 // get the current CPU frequency in kHz 2246 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2247 2248 // check if either CPU number or frequency changed 2249 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2250 mCpuNum = cpuNum; 2251 mCpukHz = cpukHz; 2252 // ignore sample for purposes of cycles 2253 valid = false; 2254 } 2255 2256 // if no change in CPU number or frequency, then record sample for cycle statistics 2257 if (valid && mCpukHz > 0) { 2258 double cycles = wcNs * cpukHz * 0.000001; 2259 mHzStats.sample(cycles); 2260 } 2261 2262 unsigned n = mWcStats.n(); 2263 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2264 if ((n & 127) == 1) { 2265 long long elapsed = mCpuUsage.elapsed(); 2266 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2267 double perLoop = elapsed / (double) n; 2268 double perLoop100 = perLoop * 0.01; 2269 double perLoop1k = perLoop * 0.001; 2270 double mean = mWcStats.mean(); 2271 double stddev = mWcStats.stddev(); 2272 double minimum = mWcStats.minimum(); 2273 double maximum = mWcStats.maximum(); 2274 double meanCycles = mHzStats.mean(); 2275 double stddevCycles = mHzStats.stddev(); 2276 double minCycles = mHzStats.minimum(); 2277 double maxCycles = mHzStats.maximum(); 2278 mCpuUsage.resetElapsed(); 2279 mWcStats.reset(); 2280 mHzStats.reset(); 2281 ALOGD("CPU usage for %s over past %.1f secs\n" 2282 " (%u mixer loops at %.1f mean ms per loop):\n" 2283 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2284 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2285 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2286 title.string(), 2287 elapsed * .000000001, n, perLoop * .000001, 2288 mean * .001, 2289 stddev * .001, 2290 minimum * .001, 2291 maximum * .001, 2292 mean / perLoop100, 2293 stddev / perLoop100, 2294 minimum / perLoop100, 2295 maximum / perLoop100, 2296 meanCycles / perLoop1k, 2297 stddevCycles / perLoop1k, 2298 minCycles / perLoop1k, 2299 maxCycles / perLoop1k); 2300 2301 } 2302 } 2303#endif 2304}; 2305 2306void AudioFlinger::PlaybackThread::checkSilentMode_l() 2307{ 2308 if (!mMasterMute) { 2309 char value[PROPERTY_VALUE_MAX]; 2310 if (property_get("ro.audio.silent", value, "0") > 0) { 2311 char *endptr; 2312 unsigned long ul = strtoul(value, &endptr, 0); 2313 if (*endptr == '\0' && ul != 0) { 2314 ALOGD("Silence is golden"); 2315 // The setprop command will not allow a property to be changed after 2316 // the first time it is set, so we don't have to worry about un-muting. 2317 setMasterMute_l(true); 2318 } 2319 } 2320 } 2321} 2322 2323bool AudioFlinger::PlaybackThread::threadLoop() 2324{ 2325 Vector< sp<Track> > tracksToRemove; 2326 2327 standbyTime = systemTime(); 2328 2329 // MIXER 2330 nsecs_t lastWarning = 0; 2331if (mType == MIXER) { 2332 longStandbyExit = false; 2333} 2334 2335 // DUPLICATING 2336 // FIXME could this be made local to while loop? 2337 writeFrames = 0; 2338 2339 cacheParameters_l(); 2340 sleepTime = idleSleepTime; 2341 2342if (mType == MIXER) { 2343 sleepTimeShift = 0; 2344} 2345 2346 CpuStats cpuStats; 2347 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2348 2349 acquireWakeLock(); 2350 2351 while (!exitPending()) 2352 { 2353 cpuStats.sample(myName); 2354 2355 Vector< sp<EffectChain> > effectChains; 2356 2357 processConfigEvents(); 2358 2359 { // scope for mLock 2360 2361 Mutex::Autolock _l(mLock); 2362 2363 if (checkForNewParameters_l()) { 2364 cacheParameters_l(); 2365 } 2366 2367 saveOutputTracks(); 2368 2369 // put audio hardware into standby after short delay 2370 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2371 mSuspended > 0)) { 2372 if (!mStandby) { 2373 2374 threadLoop_standby(); 2375 2376 mStandby = true; 2377 mBytesWritten = 0; 2378 } 2379 2380 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2381 // we're about to wait, flush the binder command buffer 2382 IPCThreadState::self()->flushCommands(); 2383 2384 clearOutputTracks(); 2385 2386 if (exitPending()) break; 2387 2388 releaseWakeLock_l(); 2389 // wait until we have something to do... 2390 ALOGV("%s going to sleep", myName.string()); 2391 mWaitWorkCV.wait(mLock); 2392 ALOGV("%s waking up", myName.string()); 2393 acquireWakeLock_l(); 2394 2395 mPrevMixerStatus = MIXER_IDLE; 2396 2397 checkSilentMode_l(); 2398 2399 standbyTime = systemTime() + standbyDelay; 2400 sleepTime = idleSleepTime; 2401 if (mType == MIXER) { 2402 sleepTimeShift = 0; 2403 } 2404 2405 continue; 2406 } 2407 } 2408 2409 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2410 // Shift in the new status; this could be a queue if it's 2411 // useful to filter the mixer status over several cycles. 2412 mPrevMixerStatus = mMixerStatus; 2413 mMixerStatus = newMixerStatus; 2414 2415 // prevent any changes in effect chain list and in each effect chain 2416 // during mixing and effect process as the audio buffers could be deleted 2417 // or modified if an effect is created or deleted 2418 lockEffectChains_l(effectChains); 2419 } 2420 2421 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2422 threadLoop_mix(); 2423 } else { 2424 threadLoop_sleepTime(); 2425 } 2426 2427 if (mSuspended > 0) { 2428 sleepTime = suspendSleepTimeUs(); 2429 } 2430 2431 // only process effects if we're going to write 2432 if (sleepTime == 0) { 2433 for (size_t i = 0; i < effectChains.size(); i ++) { 2434 effectChains[i]->process_l(); 2435 } 2436 } 2437 2438 // enable changes in effect chain 2439 unlockEffectChains(effectChains); 2440 2441 // sleepTime == 0 means we must write to audio hardware 2442 if (sleepTime == 0) { 2443 2444 threadLoop_write(); 2445 2446if (mType == MIXER) { 2447 // write blocked detection 2448 nsecs_t now = systemTime(); 2449 nsecs_t delta = now - mLastWriteTime; 2450 if (!mStandby && delta > maxPeriod) { 2451 mNumDelayedWrites++; 2452 if ((now - lastWarning) > kWarningThrottleNs) { 2453 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2454 ns2ms(delta), mNumDelayedWrites, this); 2455 lastWarning = now; 2456 } 2457 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2458 // a different threshold. Or completely removed for what it is worth anyway... 2459 if (mStandby) { 2460 longStandbyExit = true; 2461 } 2462 } 2463} 2464 2465 mStandby = false; 2466 } else { 2467 usleep(sleepTime); 2468 } 2469 2470 // Finally let go of removed track(s), without the lock held 2471 // since we can't guarantee the destructors won't acquire that 2472 // same lock. This will also mutate and push a new fast mixer state. 2473 threadLoop_removeTracks(tracksToRemove); 2474 tracksToRemove.clear(); 2475 2476 // FIXME I don't understand the need for this here; 2477 // it was in the original code but maybe the 2478 // assignment in saveOutputTracks() makes this unnecessary? 2479 clearOutputTracks(); 2480 2481 // Effect chains will be actually deleted here if they were removed from 2482 // mEffectChains list during mixing or effects processing 2483 effectChains.clear(); 2484 2485 // FIXME Note that the above .clear() is no longer necessary since effectChains 2486 // is now local to this block, but will keep it for now (at least until merge done). 2487 } 2488 2489if (mType == MIXER || mType == DIRECT) { 2490 // put output stream into standby mode 2491 if (!mStandby) { 2492 mOutput->stream->common.standby(&mOutput->stream->common); 2493 } 2494} 2495if (mType == DUPLICATING) { 2496 // for DuplicatingThread, standby mode is handled by the outputTracks 2497} 2498 2499 releaseWakeLock(); 2500 2501 ALOGV("Thread %p type %d exiting", this, mType); 2502 return false; 2503} 2504 2505// FIXME This method needs a better name. 2506// It pushes a new fast mixer state and returns (via tracksToRemove) a set of tracks to remove. 2507void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2508{ 2509 // were any of the removed tracks also fast tracks? 2510 unsigned removedMask = 0; 2511 for (size_t i = 0; i < tracksToRemove.size(); ++i) { 2512 if (tracksToRemove[i]->isFastTrack()) { 2513 int j = tracksToRemove[i]->mFastIndex; 2514 ALOG_ASSERT(0 < j && j < FastMixerState::kMaxFastTracks); 2515 removedMask |= 1 << j; 2516 } 2517 } 2518 Track* newArray[FastMixerState::kMaxFastTracks]; 2519 unsigned newMask; 2520 { 2521 AutoMutex _l(mLock); 2522 mFastTrackAvailMask |= removedMask; 2523 newMask = mFastTrackNewMask; 2524 if (newMask) { 2525 mFastTrackNewMask = 0; 2526 memcpy(newArray, mFastTrackNewArray, sizeof(mFastTrackNewArray)); 2527#if !LOG_NDEBUG 2528 memset(mFastTrackNewArray, 0, sizeof(mFastTrackNewArray)); 2529#endif 2530 } 2531 } 2532 unsigned changedMask = newMask | removedMask; 2533 // are there any newly added or removed fast tracks? 2534 if (changedMask) { 2535 2536 // This assert would be incorrect because it's theoretically possible (though unlikely) 2537 // for a track to be created and then removed within the same normal mix cycle: 2538 // ALOG_ASSERT(!(newMask & removedMask)); 2539 // The converse, of removing a track and then creating a new track at the identical slot 2540 // within the same normal mix cycle, is impossible because the slot isn't marked available. 2541 2542 // prepare a new state to push 2543 FastMixerStateQueue *sq = mFastMixer->sq(); 2544 FastMixerState *state = sq->begin(); 2545 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2546 while (changedMask) { 2547 int j = __builtin_ctz(changedMask); 2548 ALOG_ASSERT(0 < j && j < FastMixerState::kMaxFastTracks); 2549 changedMask &= ~(1 << j); 2550 FastTrack *fastTrack = &state->mFastTracks[j]; 2551 // must first do new tracks, then removed tracks, in case same track in both 2552 if (newMask & (1 << j)) { 2553 ALOG_ASSERT(!(state->mTrackMask & (1 << j))); 2554 ALOG_ASSERT(fastTrack->mBufferProvider == NULL && 2555 fastTrack->mVolumeProvider == NULL); 2556 Track *track = newArray[j]; 2557 AudioBufferProvider *abp = track; 2558 VolumeProvider *vp = track; 2559 fastTrack->mBufferProvider = abp; 2560 fastTrack->mVolumeProvider = vp; 2561 fastTrack->mSampleRate = track->mSampleRate; 2562 fastTrack->mChannelMask = track->mChannelMask; 2563 state->mTrackMask |= 1 << j; 2564 } 2565 if (removedMask & (1 << j)) { 2566 ALOG_ASSERT(state->mTrackMask & (1 << j)); 2567 ALOG_ASSERT(fastTrack->mBufferProvider != NULL && 2568 fastTrack->mVolumeProvider != NULL); 2569 fastTrack->mBufferProvider = NULL; 2570 fastTrack->mVolumeProvider = NULL; 2571 fastTrack->mSampleRate = mSampleRate; 2572 fastTrack->mChannelMask = AUDIO_CHANNEL_OUT_STEREO; 2573 state->mTrackMask &= ~(1 << j); 2574 } 2575 fastTrack->mGeneration++; 2576 } 2577 state->mFastTracksGen++; 2578 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2579 if (state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2580 state->mCommand = FastMixerState::COLD_IDLE; 2581 state->mColdFutexAddr = &mFastMixerFutex; 2582 state->mColdGen++; 2583 mFastMixerFutex = 0; 2584 mNormalSink = mOutputSink; 2585 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2586 } 2587 sq->end(); 2588 // If any fast tracks were removed, we must wait for acknowledgement 2589 // because we're about to decrement the last sp<> on those tracks. 2590 // Similarly if we put it into cold idle, need to wait for acknowledgement 2591 // so that it stops doing I/O. 2592 if (removedMask) { 2593 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2594 } 2595 sq->push(block); 2596 } 2597 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2598} 2599 2600void AudioFlinger::MixerThread::threadLoop_write() 2601{ 2602 // FIXME we should only do one push per cycle; confirm this is true 2603 // Start the fast mixer if it's not already running 2604 if (mFastMixer != NULL) { 2605 FastMixerStateQueue *sq = mFastMixer->sq(); 2606 FastMixerState *state = sq->begin(); 2607 if (state->mCommand != FastMixerState::MIX_WRITE && state->mTrackMask > 1) { 2608 if (state->mCommand == FastMixerState::COLD_IDLE) { 2609 int32_t old = android_atomic_inc(&mFastMixerFutex); 2610 if (old == -1) { 2611 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2612 } 2613 } 2614 state->mCommand = FastMixerState::MIX_WRITE; 2615 sq->end(); 2616 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2617 mNormalSink = mPipeSink; 2618 } else { 2619 sq->end(false /*didModify*/); 2620 } 2621 } 2622 PlaybackThread::threadLoop_write(); 2623} 2624 2625// shared by MIXER and DIRECT, overridden by DUPLICATING 2626void AudioFlinger::PlaybackThread::threadLoop_write() 2627{ 2628 // FIXME rewrite to reduce number of system calls 2629 mLastWriteTime = systemTime(); 2630 mInWrite = true; 2631 int bytesWritten; 2632 2633 // If an NBAIO sink is present, use it to write the normal mixer's submix 2634 if (mNormalSink != 0) { 2635#define mBitShift 2 // FIXME 2636 size_t count = mixBufferSize >> mBitShift; 2637 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2638 if (framesWritten > 0) { 2639 bytesWritten = framesWritten << mBitShift; 2640 } else { 2641 bytesWritten = framesWritten; 2642 } 2643 2644 // otherwise use the HAL / AudioStreamOut directly 2645 } else { 2646 // FIXME legacy, remove 2647 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2648 } 2649 2650 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2651 mNumWrites++; 2652 mInWrite = false; 2653} 2654 2655void AudioFlinger::MixerThread::threadLoop_standby() 2656{ 2657 // Idle the fast mixer if it's currently running 2658 if (mFastMixer != NULL) { 2659 FastMixerStateQueue *sq = mFastMixer->sq(); 2660 FastMixerState *state = sq->begin(); 2661 if (!(state->mCommand & FastMixerState::IDLE)) { 2662 state->mCommand = FastMixerState::COLD_IDLE; 2663 state->mColdFutexAddr = &mFastMixerFutex; 2664 state->mColdGen++; 2665 mFastMixerFutex = 0; 2666 sq->end(); 2667 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2668 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2669 mNormalSink = mOutputSink; 2670 } else { 2671 sq->end(false /*didModify*/); 2672 } 2673 } 2674 PlaybackThread::threadLoop_standby(); 2675} 2676 2677// shared by MIXER and DIRECT, overridden by DUPLICATING 2678void AudioFlinger::PlaybackThread::threadLoop_standby() 2679{ 2680 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2681 mOutput->stream->common.standby(&mOutput->stream->common); 2682} 2683 2684void AudioFlinger::MixerThread::threadLoop_mix() 2685{ 2686 // obtain the presentation timestamp of the next output buffer 2687 int64_t pts; 2688 status_t status = INVALID_OPERATION; 2689 2690 if (NULL != mOutput->stream->get_next_write_timestamp) { 2691 status = mOutput->stream->get_next_write_timestamp( 2692 mOutput->stream, &pts); 2693 } 2694 2695 if (status != NO_ERROR) { 2696 pts = AudioBufferProvider::kInvalidPTS; 2697 } 2698 2699 // mix buffers... 2700 mAudioMixer->process(pts); 2701 // increase sleep time progressively when application underrun condition clears. 2702 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2703 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2704 // such that we would underrun the audio HAL. 2705 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2706 sleepTimeShift--; 2707 } 2708 sleepTime = 0; 2709 standbyTime = systemTime() + standbyDelay; 2710 //TODO: delay standby when effects have a tail 2711} 2712 2713void AudioFlinger::MixerThread::threadLoop_sleepTime() 2714{ 2715 // If no tracks are ready, sleep once for the duration of an output 2716 // buffer size, then write 0s to the output 2717 if (sleepTime == 0) { 2718 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2719 sleepTime = activeSleepTime >> sleepTimeShift; 2720 if (sleepTime < kMinThreadSleepTimeUs) { 2721 sleepTime = kMinThreadSleepTimeUs; 2722 } 2723 // reduce sleep time in case of consecutive application underruns to avoid 2724 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2725 // duration we would end up writing less data than needed by the audio HAL if 2726 // the condition persists. 2727 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2728 sleepTimeShift++; 2729 } 2730 } else { 2731 sleepTime = idleSleepTime; 2732 } 2733 } else if (mBytesWritten != 0 || 2734 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2735 memset (mMixBuffer, 0, mixBufferSize); 2736 sleepTime = 0; 2737 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2738 } 2739 // TODO add standby time extension fct of effect tail 2740} 2741 2742// prepareTracks_l() must be called with ThreadBase::mLock held 2743AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2744 Vector< sp<Track> > *tracksToRemove) 2745{ 2746 2747 mixer_state mixerStatus = MIXER_IDLE; 2748 // find out which tracks need to be processed 2749 size_t count = mActiveTracks.size(); 2750 size_t mixedTracks = 0; 2751 size_t tracksWithEffect = 0; 2752 size_t fastTracks = 0; 2753 2754 float masterVolume = mMasterVolume; 2755 bool masterMute = mMasterMute; 2756 2757 if (masterMute) { 2758 masterVolume = 0; 2759 } 2760 // Delegate master volume control to effect in output mix effect chain if needed 2761 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2762 if (chain != 0) { 2763 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2764 chain->setVolume_l(&v, &v); 2765 masterVolume = (float)((v + (1 << 23)) >> 24); 2766 chain.clear(); 2767 } 2768 2769 for (size_t i=0 ; i<count ; i++) { 2770 sp<Track> t = mActiveTracks[i].promote(); 2771 if (t == 0) continue; 2772 2773 // this const just means the local variable doesn't change 2774 Track* const track = t.get(); 2775 2776 if (track->isFastTrack()) { 2777 // cache the combined master volume and stream type volume for fast mixer; 2778 // this lacks any synchronization or barrier so VolumeProvider may read a stale value 2779 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2780 ++fastTracks; 2781 if (track->isTerminated()) { 2782 tracksToRemove->add(track); 2783 } 2784 continue; 2785 } 2786 2787 { // local variable scope to avoid goto warning 2788 2789 audio_track_cblk_t* cblk = track->cblk(); 2790 2791 // The first time a track is added we wait 2792 // for all its buffers to be filled before processing it 2793 int name = track->name(); 2794 // make sure that we have enough frames to mix one full buffer. 2795 // enforce this condition only once to enable draining the buffer in case the client 2796 // app does not call stop() and relies on underrun to stop: 2797 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2798 // during last round 2799 uint32_t minFrames = 1; 2800 if (!track->isStopped() && !track->isPausing() && 2801 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2802 if (t->sampleRate() == (int)mSampleRate) { 2803 minFrames = mNormalFrameCount; 2804 } else { 2805 // +1 for rounding and +1 for additional sample needed for interpolation 2806 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2807 // add frames already consumed but not yet released by the resampler 2808 // because cblk->framesReady() will include these frames 2809 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2810 // the minimum track buffer size is normally twice the number of frames necessary 2811 // to fill one buffer and the resampler should not leave more than one buffer worth 2812 // of unreleased frames after each pass, but just in case... 2813 ALOG_ASSERT(minFrames <= cblk->frameCount); 2814 } 2815 } 2816 if ((track->framesReady() >= minFrames) && track->isReady() && 2817 !track->isPaused() && !track->isTerminated()) 2818 { 2819 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2820 2821 mixedTracks++; 2822 2823 // track->mainBuffer() != mMixBuffer means there is an effect chain 2824 // connected to the track 2825 chain.clear(); 2826 if (track->mainBuffer() != mMixBuffer) { 2827 chain = getEffectChain_l(track->sessionId()); 2828 // Delegate volume control to effect in track effect chain if needed 2829 if (chain != 0) { 2830 tracksWithEffect++; 2831 } else { 2832 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2833 name, track->sessionId()); 2834 } 2835 } 2836 2837 2838 int param = AudioMixer::VOLUME; 2839 if (track->mFillingUpStatus == Track::FS_FILLED) { 2840 // no ramp for the first volume setting 2841 track->mFillingUpStatus = Track::FS_ACTIVE; 2842 if (track->mState == TrackBase::RESUMING) { 2843 track->mState = TrackBase::ACTIVE; 2844 param = AudioMixer::RAMP_VOLUME; 2845 } 2846 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2847 } else if (cblk->server != 0) { 2848 // If the track is stopped before the first frame was mixed, 2849 // do not apply ramp 2850 param = AudioMixer::RAMP_VOLUME; 2851 } 2852 2853 // compute volume for this track 2854 uint32_t vl, vr, va; 2855 if (track->isMuted() || track->isPausing() || 2856 mStreamTypes[track->streamType()].mute) { 2857 vl = vr = va = 0; 2858 if (track->isPausing()) { 2859 track->setPaused(); 2860 } 2861 } else { 2862 2863 // read original volumes with volume control 2864 float typeVolume = mStreamTypes[track->streamType()].volume; 2865 float v = masterVolume * typeVolume; 2866 uint32_t vlr = cblk->getVolumeLR(); 2867 vl = vlr & 0xFFFF; 2868 vr = vlr >> 16; 2869 // track volumes come from shared memory, so can't be trusted and must be clamped 2870 if (vl > MAX_GAIN_INT) { 2871 ALOGV("Track left volume out of range: %04X", vl); 2872 vl = MAX_GAIN_INT; 2873 } 2874 if (vr > MAX_GAIN_INT) { 2875 ALOGV("Track right volume out of range: %04X", vr); 2876 vr = MAX_GAIN_INT; 2877 } 2878 // now apply the master volume and stream type volume 2879 vl = (uint32_t)(v * vl) << 12; 2880 vr = (uint32_t)(v * vr) << 12; 2881 // assuming master volume and stream type volume each go up to 1.0, 2882 // vl and vr are now in 8.24 format 2883 2884 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2885 // send level comes from shared memory and so may be corrupt 2886 if (sendLevel > MAX_GAIN_INT) { 2887 ALOGV("Track send level out of range: %04X", sendLevel); 2888 sendLevel = MAX_GAIN_INT; 2889 } 2890 va = (uint32_t)(v * sendLevel); 2891 } 2892 // Delegate volume control to effect in track effect chain if needed 2893 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2894 // Do not ramp volume if volume is controlled by effect 2895 param = AudioMixer::VOLUME; 2896 track->mHasVolumeController = true; 2897 } else { 2898 // force no volume ramp when volume controller was just disabled or removed 2899 // from effect chain to avoid volume spike 2900 if (track->mHasVolumeController) { 2901 param = AudioMixer::VOLUME; 2902 } 2903 track->mHasVolumeController = false; 2904 } 2905 2906 // Convert volumes from 8.24 to 4.12 format 2907 // This additional clamping is needed in case chain->setVolume_l() overshot 2908 vl = (vl + (1 << 11)) >> 12; 2909 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2910 vr = (vr + (1 << 11)) >> 12; 2911 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2912 2913 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2914 2915 // XXX: these things DON'T need to be done each time 2916 mAudioMixer->setBufferProvider(name, track); 2917 mAudioMixer->enable(name); 2918 2919 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2920 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2921 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2922 mAudioMixer->setParameter( 2923 name, 2924 AudioMixer::TRACK, 2925 AudioMixer::FORMAT, (void *)track->format()); 2926 mAudioMixer->setParameter( 2927 name, 2928 AudioMixer::TRACK, 2929 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2930 mAudioMixer->setParameter( 2931 name, 2932 AudioMixer::RESAMPLE, 2933 AudioMixer::SAMPLE_RATE, 2934 (void *)(cblk->sampleRate)); 2935 mAudioMixer->setParameter( 2936 name, 2937 AudioMixer::TRACK, 2938 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2939 mAudioMixer->setParameter( 2940 name, 2941 AudioMixer::TRACK, 2942 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2943 2944 // reset retry count 2945 track->mRetryCount = kMaxTrackRetries; 2946 2947 // If one track is ready, set the mixer ready if: 2948 // - the mixer was not ready during previous round OR 2949 // - no other track is not ready 2950 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2951 mixerStatus != MIXER_TRACKS_ENABLED) { 2952 mixerStatus = MIXER_TRACKS_READY; 2953 } 2954 } else { 2955 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2956 if (track->isStopped()) { 2957 track->reset(); 2958 } 2959 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2960 // We have consumed all the buffers of this track. 2961 // Remove it from the list of active tracks. 2962 // TODO: use actual buffer filling status instead of latency when available from 2963 // audio HAL 2964 size_t audioHALFrames = 2965 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2966 size_t framesWritten = 2967 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2968 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2969 tracksToRemove->add(track); 2970 } 2971 } else { 2972 // No buffers for this track. Give it a few chances to 2973 // fill a buffer, then remove it from active list. 2974 if (--(track->mRetryCount) <= 0) { 2975 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2976 tracksToRemove->add(track); 2977 // indicate to client process that the track was disabled because of underrun 2978 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2979 // If one track is not ready, mark the mixer also not ready if: 2980 // - the mixer was ready during previous round OR 2981 // - no other track is ready 2982 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2983 mixerStatus != MIXER_TRACKS_READY) { 2984 mixerStatus = MIXER_TRACKS_ENABLED; 2985 } 2986 } 2987 mAudioMixer->disable(name); 2988 } 2989 2990 } // local variable scope to avoid goto warning 2991track_is_ready: ; 2992 2993 } 2994 2995 // FIXME Here is where we would push the new FastMixer state if necessary 2996 2997 // remove all the tracks that need to be... 2998 count = tracksToRemove->size(); 2999 if (CC_UNLIKELY(count)) { 3000 for (size_t i=0 ; i<count ; i++) { 3001 const sp<Track>& track = tracksToRemove->itemAt(i); 3002 mActiveTracks.remove(track); 3003 if (track->mainBuffer() != mMixBuffer) { 3004 chain = getEffectChain_l(track->sessionId()); 3005 if (chain != 0) { 3006 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3007 chain->decActiveTrackCnt(); 3008 } 3009 } 3010 if (track->isTerminated()) { 3011 removeTrack_l(track); 3012 } 3013 } 3014 } 3015 3016 // mix buffer must be cleared if all tracks are connected to an 3017 // effect chain as in this case the mixer will not write to 3018 // mix buffer and track effects will accumulate into it 3019 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3020 // FIXME as a performance optimization, should remember previous zero status 3021 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3022 } 3023 3024 // if any fast tracks, then status is ready 3025 if (fastTracks > 0) { 3026 mixerStatus = MIXER_TRACKS_READY; 3027 } 3028 return mixerStatus; 3029} 3030 3031/* 3032The derived values that are cached: 3033 - mixBufferSize from frame count * frame size 3034 - activeSleepTime from activeSleepTimeUs() 3035 - idleSleepTime from idleSleepTimeUs() 3036 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3037 - maxPeriod from frame count and sample rate (MIXER only) 3038 3039The parameters that affect these derived values are: 3040 - frame count 3041 - frame size 3042 - sample rate 3043 - device type: A2DP or not 3044 - device latency 3045 - format: PCM or not 3046 - active sleep time 3047 - idle sleep time 3048*/ 3049 3050void AudioFlinger::PlaybackThread::cacheParameters_l() 3051{ 3052 mixBufferSize = mNormalFrameCount * mFrameSize; 3053 activeSleepTime = activeSleepTimeUs(); 3054 idleSleepTime = idleSleepTimeUs(); 3055} 3056 3057void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3058{ 3059 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3060 this, streamType, mTracks.size()); 3061 Mutex::Autolock _l(mLock); 3062 3063 size_t size = mTracks.size(); 3064 for (size_t i = 0; i < size; i++) { 3065 sp<Track> t = mTracks[i]; 3066 if (t->streamType() == streamType) { 3067 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3068 t->mCblk->cv.signal(); 3069 } 3070 } 3071} 3072 3073// getTrackName_l() must be called with ThreadBase::mLock held 3074int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3075{ 3076 return mAudioMixer->getTrackName(channelMask); 3077} 3078 3079// deleteTrackName_l() must be called with ThreadBase::mLock held 3080void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3081{ 3082 ALOGV("remove track (%d) and delete from mixer", name); 3083 mAudioMixer->deleteTrackName(name); 3084} 3085 3086// checkForNewParameters_l() must be called with ThreadBase::mLock held 3087bool AudioFlinger::MixerThread::checkForNewParameters_l() 3088{ 3089 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3090 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3091 bool reconfig = false; 3092 3093 while (!mNewParameters.isEmpty()) { 3094 3095 if (mFastMixer != NULL) { 3096 FastMixerStateQueue *sq = mFastMixer->sq(); 3097 FastMixerState *state = sq->begin(); 3098 if (!(state->mCommand & FastMixerState::IDLE)) { 3099 previousCommand = state->mCommand; 3100 state->mCommand = FastMixerState::HOT_IDLE; 3101 sq->end(); 3102 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3103 } else { 3104 sq->end(false /*didModify*/); 3105 } 3106 } 3107 3108 status_t status = NO_ERROR; 3109 String8 keyValuePair = mNewParameters[0]; 3110 AudioParameter param = AudioParameter(keyValuePair); 3111 int value; 3112 3113 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3114 reconfig = true; 3115 } 3116 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3117 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3118 status = BAD_VALUE; 3119 } else { 3120 reconfig = true; 3121 } 3122 } 3123 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3124 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3125 status = BAD_VALUE; 3126 } else { 3127 reconfig = true; 3128 } 3129 } 3130 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3131 // do not accept frame count changes if tracks are open as the track buffer 3132 // size depends on frame count and correct behavior would not be guaranteed 3133 // if frame count is changed after track creation 3134 if (!mTracks.isEmpty()) { 3135 status = INVALID_OPERATION; 3136 } else { 3137 reconfig = true; 3138 } 3139 } 3140 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3141#ifdef ADD_BATTERY_DATA 3142 // when changing the audio output device, call addBatteryData to notify 3143 // the change 3144 if ((int)mDevice != value) { 3145 uint32_t params = 0; 3146 // check whether speaker is on 3147 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3148 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3149 } 3150 3151 int deviceWithoutSpeaker 3152 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3153 // check if any other device (except speaker) is on 3154 if (value & deviceWithoutSpeaker ) { 3155 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3156 } 3157 3158 if (params != 0) { 3159 addBatteryData(params); 3160 } 3161 } 3162#endif 3163 3164 // forward device change to effects that have requested to be 3165 // aware of attached audio device. 3166 mDevice = (uint32_t)value; 3167 for (size_t i = 0; i < mEffectChains.size(); i++) { 3168 mEffectChains[i]->setDevice_l(mDevice); 3169 } 3170 } 3171 3172 if (status == NO_ERROR) { 3173 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3174 keyValuePair.string()); 3175 if (!mStandby && status == INVALID_OPERATION) { 3176 mOutput->stream->common.standby(&mOutput->stream->common); 3177 mStandby = true; 3178 mBytesWritten = 0; 3179 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3180 keyValuePair.string()); 3181 } 3182 if (status == NO_ERROR && reconfig) { 3183 delete mAudioMixer; 3184 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3185 mAudioMixer = NULL; 3186 readOutputParameters(); 3187 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3188 for (size_t i = 0; i < mTracks.size() ; i++) { 3189 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3190 if (name < 0) break; 3191 mTracks[i]->mName = name; 3192 // limit track sample rate to 2 x new output sample rate 3193 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3194 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3195 } 3196 } 3197 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3198 } 3199 } 3200 3201 mNewParameters.removeAt(0); 3202 3203 mParamStatus = status; 3204 mParamCond.signal(); 3205 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3206 // already timed out waiting for the status and will never signal the condition. 3207 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3208 } 3209 3210 if (!(previousCommand & FastMixerState::IDLE)) { 3211 ALOG_ASSERT(mFastMixer != NULL); 3212 FastMixerStateQueue *sq = mFastMixer->sq(); 3213 FastMixerState *state = sq->begin(); 3214 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3215 state->mCommand = previousCommand; 3216 sq->end(); 3217 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3218 } 3219 3220 return reconfig; 3221} 3222 3223status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3224{ 3225 const size_t SIZE = 256; 3226 char buffer[SIZE]; 3227 String8 result; 3228 3229 PlaybackThread::dumpInternals(fd, args); 3230 3231 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3232 result.append(buffer); 3233 write(fd, result.string(), result.size()); 3234 3235 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3236 FastMixerDumpState copy = mFastMixerDumpState; 3237 copy.dump(fd); 3238 3239 return NO_ERROR; 3240} 3241 3242uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3243{ 3244 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3245} 3246 3247uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3248{ 3249 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3250} 3251 3252void AudioFlinger::MixerThread::cacheParameters_l() 3253{ 3254 PlaybackThread::cacheParameters_l(); 3255 3256 // FIXME: Relaxed timing because of a certain device that can't meet latency 3257 // Should be reduced to 2x after the vendor fixes the driver issue 3258 // increase threshold again due to low power audio mode. The way this warning 3259 // threshold is calculated and its usefulness should be reconsidered anyway. 3260 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3261} 3262 3263// ---------------------------------------------------------------------------- 3264AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3265 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3266 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3267 // mLeftVolFloat, mRightVolFloat 3268 // mLeftVolShort, mRightVolShort 3269{ 3270} 3271 3272AudioFlinger::DirectOutputThread::~DirectOutputThread() 3273{ 3274} 3275 3276AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3277 Vector< sp<Track> > *tracksToRemove 3278) 3279{ 3280 sp<Track> trackToRemove; 3281 3282 mixer_state mixerStatus = MIXER_IDLE; 3283 3284 // find out which tracks need to be processed 3285 if (mActiveTracks.size() != 0) { 3286 sp<Track> t = mActiveTracks[0].promote(); 3287 // The track died recently 3288 if (t == 0) return MIXER_IDLE; 3289 3290 Track* const track = t.get(); 3291 audio_track_cblk_t* cblk = track->cblk(); 3292 3293 // The first time a track is added we wait 3294 // for all its buffers to be filled before processing it 3295 if (cblk->framesReady() && track->isReady() && 3296 !track->isPaused() && !track->isTerminated()) 3297 { 3298 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3299 3300 if (track->mFillingUpStatus == Track::FS_FILLED) { 3301 track->mFillingUpStatus = Track::FS_ACTIVE; 3302 mLeftVolFloat = mRightVolFloat = 0; 3303 mLeftVolShort = mRightVolShort = 0; 3304 if (track->mState == TrackBase::RESUMING) { 3305 track->mState = TrackBase::ACTIVE; 3306 rampVolume = true; 3307 } 3308 } else if (cblk->server != 0) { 3309 // If the track is stopped before the first frame was mixed, 3310 // do not apply ramp 3311 rampVolume = true; 3312 } 3313 // compute volume for this track 3314 float left, right; 3315 if (track->isMuted() || mMasterMute || track->isPausing() || 3316 mStreamTypes[track->streamType()].mute) { 3317 left = right = 0; 3318 if (track->isPausing()) { 3319 track->setPaused(); 3320 } 3321 } else { 3322 float typeVolume = mStreamTypes[track->streamType()].volume; 3323 float v = mMasterVolume * typeVolume; 3324 uint32_t vlr = cblk->getVolumeLR(); 3325 float v_clamped = v * (vlr & 0xFFFF); 3326 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3327 left = v_clamped/MAX_GAIN; 3328 v_clamped = v * (vlr >> 16); 3329 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3330 right = v_clamped/MAX_GAIN; 3331 } 3332 3333 if (left != mLeftVolFloat || right != mRightVolFloat) { 3334 mLeftVolFloat = left; 3335 mRightVolFloat = right; 3336 3337 // If audio HAL implements volume control, 3338 // force software volume to nominal value 3339 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3340 left = 1.0f; 3341 right = 1.0f; 3342 } 3343 3344 // Convert volumes from float to 8.24 3345 uint32_t vl = (uint32_t)(left * (1 << 24)); 3346 uint32_t vr = (uint32_t)(right * (1 << 24)); 3347 3348 // Delegate volume control to effect in track effect chain if needed 3349 // only one effect chain can be present on DirectOutputThread, so if 3350 // there is one, the track is connected to it 3351 if (!mEffectChains.isEmpty()) { 3352 // Do not ramp volume if volume is controlled by effect 3353 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3354 rampVolume = false; 3355 } 3356 } 3357 3358 // Convert volumes from 8.24 to 4.12 format 3359 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3360 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3361 leftVol = (uint16_t)v_clamped; 3362 v_clamped = (vr + (1 << 11)) >> 12; 3363 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3364 rightVol = (uint16_t)v_clamped; 3365 } else { 3366 leftVol = mLeftVolShort; 3367 rightVol = mRightVolShort; 3368 rampVolume = false; 3369 } 3370 3371 // reset retry count 3372 track->mRetryCount = kMaxTrackRetriesDirect; 3373 mActiveTrack = t; 3374 mixerStatus = MIXER_TRACKS_READY; 3375 } else { 3376 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3377 if (track->isStopped()) { 3378 track->reset(); 3379 } 3380 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3381 // We have consumed all the buffers of this track. 3382 // Remove it from the list of active tracks. 3383 // TODO: implement behavior for compressed audio 3384 size_t audioHALFrames = 3385 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3386 size_t framesWritten = 3387 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3388 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3389 trackToRemove = track; 3390 } 3391 } else { 3392 // No buffers for this track. Give it a few chances to 3393 // fill a buffer, then remove it from active list. 3394 if (--(track->mRetryCount) <= 0) { 3395 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3396 trackToRemove = track; 3397 } else { 3398 mixerStatus = MIXER_TRACKS_ENABLED; 3399 } 3400 } 3401 } 3402 } 3403 3404 // FIXME merge this with similar code for removing multiple tracks 3405 // remove all the tracks that need to be... 3406 if (CC_UNLIKELY(trackToRemove != 0)) { 3407 tracksToRemove->add(trackToRemove); 3408 mActiveTracks.remove(trackToRemove); 3409 if (!mEffectChains.isEmpty()) { 3410 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3411 trackToRemove->sessionId()); 3412 mEffectChains[0]->decActiveTrackCnt(); 3413 } 3414 if (trackToRemove->isTerminated()) { 3415 removeTrack_l(trackToRemove); 3416 } 3417 } 3418 3419 return mixerStatus; 3420} 3421 3422void AudioFlinger::DirectOutputThread::threadLoop_mix() 3423{ 3424 AudioBufferProvider::Buffer buffer; 3425 size_t frameCount = mFrameCount; 3426 int8_t *curBuf = (int8_t *)mMixBuffer; 3427 // output audio to hardware 3428 while (frameCount) { 3429 buffer.frameCount = frameCount; 3430 mActiveTrack->getNextBuffer(&buffer); 3431 if (CC_UNLIKELY(buffer.raw == NULL)) { 3432 memset(curBuf, 0, frameCount * mFrameSize); 3433 break; 3434 } 3435 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3436 frameCount -= buffer.frameCount; 3437 curBuf += buffer.frameCount * mFrameSize; 3438 mActiveTrack->releaseBuffer(&buffer); 3439 } 3440 sleepTime = 0; 3441 standbyTime = systemTime() + standbyDelay; 3442 mActiveTrack.clear(); 3443 3444 // apply volume 3445 3446 // Do not apply volume on compressed audio 3447 if (!audio_is_linear_pcm(mFormat)) { 3448 return; 3449 } 3450 3451 // convert to signed 16 bit before volume calculation 3452 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3453 size_t count = mFrameCount * mChannelCount; 3454 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3455 int16_t *dst = mMixBuffer + count-1; 3456 while (count--) { 3457 *dst-- = (int16_t)(*src--^0x80) << 8; 3458 } 3459 } 3460 3461 frameCount = mFrameCount; 3462 int16_t *out = mMixBuffer; 3463 if (rampVolume) { 3464 if (mChannelCount == 1) { 3465 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3466 int32_t vlInc = d / (int32_t)frameCount; 3467 int32_t vl = ((int32_t)mLeftVolShort << 16); 3468 do { 3469 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3470 out++; 3471 vl += vlInc; 3472 } while (--frameCount); 3473 3474 } else { 3475 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3476 int32_t vlInc = d / (int32_t)frameCount; 3477 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3478 int32_t vrInc = d / (int32_t)frameCount; 3479 int32_t vl = ((int32_t)mLeftVolShort << 16); 3480 int32_t vr = ((int32_t)mRightVolShort << 16); 3481 do { 3482 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3483 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3484 out += 2; 3485 vl += vlInc; 3486 vr += vrInc; 3487 } while (--frameCount); 3488 } 3489 } else { 3490 if (mChannelCount == 1) { 3491 do { 3492 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3493 out++; 3494 } while (--frameCount); 3495 } else { 3496 do { 3497 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3498 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3499 out += 2; 3500 } while (--frameCount); 3501 } 3502 } 3503 3504 // convert back to unsigned 8 bit after volume calculation 3505 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3506 size_t count = mFrameCount * mChannelCount; 3507 int16_t *src = mMixBuffer; 3508 uint8_t *dst = (uint8_t *)mMixBuffer; 3509 while (count--) { 3510 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3511 } 3512 } 3513 3514 mLeftVolShort = leftVol; 3515 mRightVolShort = rightVol; 3516} 3517 3518void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3519{ 3520 if (sleepTime == 0) { 3521 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3522 sleepTime = activeSleepTime; 3523 } else { 3524 sleepTime = idleSleepTime; 3525 } 3526 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3527 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3528 sleepTime = 0; 3529 } 3530} 3531 3532// getTrackName_l() must be called with ThreadBase::mLock held 3533int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3534{ 3535 return 0; 3536} 3537 3538// deleteTrackName_l() must be called with ThreadBase::mLock held 3539void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3540{ 3541} 3542 3543// checkForNewParameters_l() must be called with ThreadBase::mLock held 3544bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3545{ 3546 bool reconfig = false; 3547 3548 while (!mNewParameters.isEmpty()) { 3549 status_t status = NO_ERROR; 3550 String8 keyValuePair = mNewParameters[0]; 3551 AudioParameter param = AudioParameter(keyValuePair); 3552 int value; 3553 3554 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3555 // do not accept frame count changes if tracks are open as the track buffer 3556 // size depends on frame count and correct behavior would not be garantied 3557 // if frame count is changed after track creation 3558 if (!mTracks.isEmpty()) { 3559 status = INVALID_OPERATION; 3560 } else { 3561 reconfig = true; 3562 } 3563 } 3564 if (status == NO_ERROR) { 3565 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3566 keyValuePair.string()); 3567 if (!mStandby && status == INVALID_OPERATION) { 3568 mOutput->stream->common.standby(&mOutput->stream->common); 3569 mStandby = true; 3570 mBytesWritten = 0; 3571 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3572 keyValuePair.string()); 3573 } 3574 if (status == NO_ERROR && reconfig) { 3575 readOutputParameters(); 3576 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3577 } 3578 } 3579 3580 mNewParameters.removeAt(0); 3581 3582 mParamStatus = status; 3583 mParamCond.signal(); 3584 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3585 // already timed out waiting for the status and will never signal the condition. 3586 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3587 } 3588 return reconfig; 3589} 3590 3591uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3592{ 3593 uint32_t time; 3594 if (audio_is_linear_pcm(mFormat)) { 3595 time = PlaybackThread::activeSleepTimeUs(); 3596 } else { 3597 time = 10000; 3598 } 3599 return time; 3600} 3601 3602uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3603{ 3604 uint32_t time; 3605 if (audio_is_linear_pcm(mFormat)) { 3606 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3607 } else { 3608 time = 10000; 3609 } 3610 return time; 3611} 3612 3613uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3614{ 3615 uint32_t time; 3616 if (audio_is_linear_pcm(mFormat)) { 3617 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3618 } else { 3619 time = 10000; 3620 } 3621 return time; 3622} 3623 3624void AudioFlinger::DirectOutputThread::cacheParameters_l() 3625{ 3626 PlaybackThread::cacheParameters_l(); 3627 3628 // use shorter standby delay as on normal output to release 3629 // hardware resources as soon as possible 3630 standbyDelay = microseconds(activeSleepTime*2); 3631} 3632 3633// ---------------------------------------------------------------------------- 3634 3635AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3636 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3637 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3638 mWaitTimeMs(UINT_MAX) 3639{ 3640 addOutputTrack(mainThread); 3641} 3642 3643AudioFlinger::DuplicatingThread::~DuplicatingThread() 3644{ 3645 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3646 mOutputTracks[i]->destroy(); 3647 } 3648} 3649 3650void AudioFlinger::DuplicatingThread::threadLoop_mix() 3651{ 3652 // mix buffers... 3653 if (outputsReady(outputTracks)) { 3654 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3655 } else { 3656 memset(mMixBuffer, 0, mixBufferSize); 3657 } 3658 sleepTime = 0; 3659 writeFrames = mNormalFrameCount; 3660} 3661 3662void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3663{ 3664 if (sleepTime == 0) { 3665 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3666 sleepTime = activeSleepTime; 3667 } else { 3668 sleepTime = idleSleepTime; 3669 } 3670 } else if (mBytesWritten != 0) { 3671 // flush remaining overflow buffers in output tracks 3672 for (size_t i = 0; i < outputTracks.size(); i++) { 3673 if (outputTracks[i]->isActive()) { 3674 sleepTime = 0; 3675 writeFrames = 0; 3676 memset(mMixBuffer, 0, mixBufferSize); 3677 break; 3678 } 3679 } 3680 } 3681} 3682 3683void AudioFlinger::DuplicatingThread::threadLoop_write() 3684{ 3685 standbyTime = systemTime() + standbyDelay; 3686 for (size_t i = 0; i < outputTracks.size(); i++) { 3687 outputTracks[i]->write(mMixBuffer, writeFrames); 3688 } 3689 mBytesWritten += mixBufferSize; 3690} 3691 3692void AudioFlinger::DuplicatingThread::threadLoop_standby() 3693{ 3694 // DuplicatingThread implements standby by stopping all tracks 3695 for (size_t i = 0; i < outputTracks.size(); i++) { 3696 outputTracks[i]->stop(); 3697 } 3698} 3699 3700void AudioFlinger::DuplicatingThread::saveOutputTracks() 3701{ 3702 outputTracks = mOutputTracks; 3703} 3704 3705void AudioFlinger::DuplicatingThread::clearOutputTracks() 3706{ 3707 outputTracks.clear(); 3708} 3709 3710void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3711{ 3712 Mutex::Autolock _l(mLock); 3713 // FIXME explain this formula 3714 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3715 OutputTrack *outputTrack = new OutputTrack(thread, 3716 this, 3717 mSampleRate, 3718 mFormat, 3719 mChannelMask, 3720 frameCount); 3721 if (outputTrack->cblk() != NULL) { 3722 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3723 mOutputTracks.add(outputTrack); 3724 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3725 updateWaitTime_l(); 3726 } 3727} 3728 3729void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3730{ 3731 Mutex::Autolock _l(mLock); 3732 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3733 if (mOutputTracks[i]->thread() == thread) { 3734 mOutputTracks[i]->destroy(); 3735 mOutputTracks.removeAt(i); 3736 updateWaitTime_l(); 3737 return; 3738 } 3739 } 3740 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3741} 3742 3743// caller must hold mLock 3744void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3745{ 3746 mWaitTimeMs = UINT_MAX; 3747 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3748 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3749 if (strong != 0) { 3750 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3751 if (waitTimeMs < mWaitTimeMs) { 3752 mWaitTimeMs = waitTimeMs; 3753 } 3754 } 3755 } 3756} 3757 3758 3759bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3760{ 3761 for (size_t i = 0; i < outputTracks.size(); i++) { 3762 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3763 if (thread == 0) { 3764 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3765 return false; 3766 } 3767 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3768 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3769 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3770 return false; 3771 } 3772 } 3773 return true; 3774} 3775 3776uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3777{ 3778 return (mWaitTimeMs * 1000) / 2; 3779} 3780 3781void AudioFlinger::DuplicatingThread::cacheParameters_l() 3782{ 3783 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3784 updateWaitTime_l(); 3785 3786 MixerThread::cacheParameters_l(); 3787} 3788 3789// ---------------------------------------------------------------------------- 3790 3791// TrackBase constructor must be called with AudioFlinger::mLock held 3792AudioFlinger::ThreadBase::TrackBase::TrackBase( 3793 ThreadBase *thread, 3794 const sp<Client>& client, 3795 uint32_t sampleRate, 3796 audio_format_t format, 3797 uint32_t channelMask, 3798 int frameCount, 3799 const sp<IMemory>& sharedBuffer, 3800 int sessionId) 3801 : RefBase(), 3802 mThread(thread), 3803 mClient(client), 3804 mCblk(NULL), 3805 // mBuffer 3806 // mBufferEnd 3807 mFrameCount(0), 3808 mState(IDLE), 3809 mSampleRate(sampleRate), 3810 mFormat(format), 3811 mStepServerFailed(false), 3812 mSessionId(sessionId) 3813 // mChannelCount 3814 // mChannelMask 3815{ 3816 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3817 3818 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3819 size_t size = sizeof(audio_track_cblk_t); 3820 uint8_t channelCount = popcount(channelMask); 3821 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3822 if (sharedBuffer == 0) { 3823 size += bufferSize; 3824 } 3825 3826 if (client != NULL) { 3827 mCblkMemory = client->heap()->allocate(size); 3828 if (mCblkMemory != 0) { 3829 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3830 if (mCblk != NULL) { // construct the shared structure in-place. 3831 new(mCblk) audio_track_cblk_t(); 3832 // clear all buffers 3833 mCblk->frameCount = frameCount; 3834 mCblk->sampleRate = sampleRate; 3835// uncomment the following lines to quickly test 32-bit wraparound 3836// mCblk->user = 0xffff0000; 3837// mCblk->server = 0xffff0000; 3838// mCblk->userBase = 0xffff0000; 3839// mCblk->serverBase = 0xffff0000; 3840 mChannelCount = channelCount; 3841 mChannelMask = channelMask; 3842 if (sharedBuffer == 0) { 3843 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3844 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3845 // Force underrun condition to avoid false underrun callback until first data is 3846 // written to buffer (other flags are cleared) 3847 mCblk->flags = CBLK_UNDERRUN_ON; 3848 } else { 3849 mBuffer = sharedBuffer->pointer(); 3850 } 3851 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3852 } 3853 } else { 3854 ALOGE("not enough memory for AudioTrack size=%u", size); 3855 client->heap()->dump("AudioTrack"); 3856 return; 3857 } 3858 } else { 3859 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3860 // construct the shared structure in-place. 3861 new(mCblk) audio_track_cblk_t(); 3862 // clear all buffers 3863 mCblk->frameCount = frameCount; 3864 mCblk->sampleRate = sampleRate; 3865// uncomment the following lines to quickly test 32-bit wraparound 3866// mCblk->user = 0xffff0000; 3867// mCblk->server = 0xffff0000; 3868// mCblk->userBase = 0xffff0000; 3869// mCblk->serverBase = 0xffff0000; 3870 mChannelCount = channelCount; 3871 mChannelMask = channelMask; 3872 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3873 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3874 // Force underrun condition to avoid false underrun callback until first data is 3875 // written to buffer (other flags are cleared) 3876 mCblk->flags = CBLK_UNDERRUN_ON; 3877 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3878 } 3879} 3880 3881AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3882{ 3883 if (mCblk != NULL) { 3884 if (mClient == 0) { 3885 delete mCblk; 3886 } else { 3887 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3888 } 3889 } 3890 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3891 if (mClient != 0) { 3892 // Client destructor must run with AudioFlinger mutex locked 3893 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3894 // If the client's reference count drops to zero, the associated destructor 3895 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3896 // relying on the automatic clear() at end of scope. 3897 mClient.clear(); 3898 } 3899} 3900 3901// AudioBufferProvider interface 3902// getNextBuffer() = 0; 3903// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3904void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3905{ 3906 buffer->raw = NULL; 3907 mFrameCount = buffer->frameCount; 3908 (void) step(); // ignore return value of step() 3909 buffer->frameCount = 0; 3910} 3911 3912bool AudioFlinger::ThreadBase::TrackBase::step() { 3913 bool result; 3914 audio_track_cblk_t* cblk = this->cblk(); 3915 3916 result = cblk->stepServer(mFrameCount); 3917 if (!result) { 3918 ALOGV("stepServer failed acquiring cblk mutex"); 3919 mStepServerFailed = true; 3920 } 3921 return result; 3922} 3923 3924void AudioFlinger::ThreadBase::TrackBase::reset() { 3925 audio_track_cblk_t* cblk = this->cblk(); 3926 3927 cblk->user = 0; 3928 cblk->server = 0; 3929 cblk->userBase = 0; 3930 cblk->serverBase = 0; 3931 mStepServerFailed = false; 3932 ALOGV("TrackBase::reset"); 3933} 3934 3935int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3936 return (int)mCblk->sampleRate; 3937} 3938 3939void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3940 audio_track_cblk_t* cblk = this->cblk(); 3941 size_t frameSize = cblk->frameSize; 3942 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3943 int8_t *bufferEnd = bufferStart + frames * frameSize; 3944 3945 // Check validity of returned pointer in case the track control block would have been corrupted. 3946 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 3947 "TrackBase::getBuffer buffer out of range:\n" 3948 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 3949 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 3950 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3951 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 3952 3953 return bufferStart; 3954} 3955 3956status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3957{ 3958 mSyncEvents.add(event); 3959 return NO_ERROR; 3960} 3961 3962// ---------------------------------------------------------------------------- 3963 3964// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3965AudioFlinger::PlaybackThread::Track::Track( 3966 PlaybackThread *thread, 3967 const sp<Client>& client, 3968 audio_stream_type_t streamType, 3969 uint32_t sampleRate, 3970 audio_format_t format, 3971 uint32_t channelMask, 3972 int frameCount, 3973 const sp<IMemory>& sharedBuffer, 3974 int sessionId, 3975 IAudioFlinger::track_flags_t flags) 3976 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3977 mMute(false), 3978 mFillingUpStatus(FS_INVALID), 3979 // mRetryCount initialized later when needed 3980 mSharedBuffer(sharedBuffer), 3981 mStreamType(streamType), 3982 mName(-1), // see note below 3983 mMainBuffer(thread->mixBuffer()), 3984 mAuxBuffer(NULL), 3985 mAuxEffectId(0), mHasVolumeController(false), 3986 mPresentationCompleteFrames(0), 3987 mFlags(flags), 3988 mFastIndex(-1), 3989 mCachedVolume(1.0) 3990{ 3991 if (mCblk != NULL) { 3992 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3993 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3994 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3995 if (flags & IAudioFlinger::TRACK_FAST) { 3996 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 3997 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 3998 int i = __builtin_ctz(thread->mFastTrackAvailMask); 3999 ALOG_ASSERT(0 < i && i < FastMixerState::kMaxFastTracks); 4000 mFastIndex = i; 4001 thread->mFastTrackAvailMask &= ~(1 << i); 4002 // Although we've allocated an index, we can't mutate or push a new fast track state 4003 // here, because that data structure can only be changed within the normal mixer 4004 // threadLoop(). So instead, make a note to mutate and push later. 4005 thread->mFastTrackNewArray[i] = this; 4006 thread->mFastTrackNewMask |= 1 << i; 4007 } 4008 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4009 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4010 if (mName < 0) { 4011 ALOGE("no more track names available"); 4012 } 4013 } 4014 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4015} 4016 4017AudioFlinger::PlaybackThread::Track::~Track() 4018{ 4019 ALOGV("PlaybackThread::Track destructor"); 4020 sp<ThreadBase> thread = mThread.promote(); 4021 if (thread != 0) { 4022 Mutex::Autolock _l(thread->mLock); 4023 mState = TERMINATED; 4024 } 4025} 4026 4027void AudioFlinger::PlaybackThread::Track::destroy() 4028{ 4029 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4030 // by removing it from mTracks vector, so there is a risk that this Tracks's 4031 // destructor is called. As the destructor needs to lock mLock, 4032 // we must acquire a strong reference on this Track before locking mLock 4033 // here so that the destructor is called only when exiting this function. 4034 // On the other hand, as long as Track::destroy() is only called by 4035 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4036 // this Track with its member mTrack. 4037 sp<Track> keep(this); 4038 { // scope for mLock 4039 sp<ThreadBase> thread = mThread.promote(); 4040 if (thread != 0) { 4041 if (!isOutputTrack()) { 4042 if (mState == ACTIVE || mState == RESUMING) { 4043 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4044 4045#ifdef ADD_BATTERY_DATA 4046 // to track the speaker usage 4047 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4048#endif 4049 } 4050 AudioSystem::releaseOutput(thread->id()); 4051 } 4052 Mutex::Autolock _l(thread->mLock); 4053 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4054 playbackThread->destroyTrack_l(this); 4055 } 4056 } 4057} 4058 4059void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4060{ 4061 uint32_t vlr = mCblk->getVolumeLR(); 4062 if (isFastTrack()) { 4063 strcpy(buffer, " fast"); 4064 } else { 4065 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4066 } 4067 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %1d %1d %1d %5u %5.2g %5.2g 0x%08x 0x%08x 0x%08x 0x%08x\n", 4068 (mClient == 0) ? getpid_cached : mClient->pid(), 4069 mStreamType, 4070 mFormat, 4071 mChannelMask, 4072 mSessionId, 4073 mFrameCount, 4074 mState, 4075 mMute, 4076 mFillingUpStatus, 4077 mCblk->sampleRate, 4078 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4079 20.0 * log10((vlr >> 16) / 4096.0), 4080 mCblk->server, 4081 mCblk->user, 4082 (int)mMainBuffer, 4083 (int)mAuxBuffer); 4084} 4085 4086// AudioBufferProvider interface 4087status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4088 AudioBufferProvider::Buffer* buffer, int64_t pts) 4089{ 4090 audio_track_cblk_t* cblk = this->cblk(); 4091 uint32_t framesReady; 4092 uint32_t framesReq = buffer->frameCount; 4093 4094 // Check if last stepServer failed, try to step now 4095 if (mStepServerFailed) { 4096 if (!step()) goto getNextBuffer_exit; 4097 ALOGV("stepServer recovered"); 4098 mStepServerFailed = false; 4099 } 4100 4101 framesReady = cblk->framesReady(); 4102 4103 if (CC_LIKELY(framesReady)) { 4104 uint32_t s = cblk->server; 4105 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4106 4107 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4108 if (framesReq > framesReady) { 4109 framesReq = framesReady; 4110 } 4111 if (framesReq > bufferEnd - s) { 4112 framesReq = bufferEnd - s; 4113 } 4114 4115 buffer->raw = getBuffer(s, framesReq); 4116 if (buffer->raw == NULL) goto getNextBuffer_exit; 4117 4118 buffer->frameCount = framesReq; 4119 return NO_ERROR; 4120 } 4121 4122getNextBuffer_exit: 4123 buffer->raw = NULL; 4124 buffer->frameCount = 0; 4125 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4126 return NOT_ENOUGH_DATA; 4127} 4128 4129uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4130 return mCblk->framesReady(); 4131} 4132 4133bool AudioFlinger::PlaybackThread::Track::isReady() const { 4134 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4135 4136 if (framesReady() >= mCblk->frameCount || 4137 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4138 mFillingUpStatus = FS_FILLED; 4139 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4140 return true; 4141 } 4142 return false; 4143} 4144 4145status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4146 int triggerSession) 4147{ 4148 status_t status = NO_ERROR; 4149 ALOGV("start(%d), calling pid %d session %d", 4150 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4151 4152 sp<ThreadBase> thread = mThread.promote(); 4153 if (thread != 0) { 4154 Mutex::Autolock _l(thread->mLock); 4155 track_state state = mState; 4156 // here the track could be either new, or restarted 4157 // in both cases "unstop" the track 4158 if (mState == PAUSED) { 4159 mState = TrackBase::RESUMING; 4160 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4161 } else { 4162 mState = TrackBase::ACTIVE; 4163 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4164 } 4165 4166 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4167 thread->mLock.unlock(); 4168 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4169 thread->mLock.lock(); 4170 4171#ifdef ADD_BATTERY_DATA 4172 // to track the speaker usage 4173 if (status == NO_ERROR) { 4174 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4175 } 4176#endif 4177 } 4178 if (status == NO_ERROR) { 4179 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4180 playbackThread->addTrack_l(this); 4181 } else { 4182 mState = state; 4183 } 4184 } else { 4185 status = BAD_VALUE; 4186 } 4187 return status; 4188} 4189 4190void AudioFlinger::PlaybackThread::Track::stop() 4191{ 4192 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4193 sp<ThreadBase> thread = mThread.promote(); 4194 if (thread != 0) { 4195 Mutex::Autolock _l(thread->mLock); 4196 track_state state = mState; 4197 if (mState > STOPPED) { 4198 mState = STOPPED; 4199 // If the track is not active (PAUSED and buffers full), flush buffers 4200 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4201 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4202 reset(); 4203 } 4204 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 4205 } 4206 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4207 thread->mLock.unlock(); 4208 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4209 thread->mLock.lock(); 4210 4211#ifdef ADD_BATTERY_DATA 4212 // to track the speaker usage 4213 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4214#endif 4215 } 4216 } 4217} 4218 4219void AudioFlinger::PlaybackThread::Track::pause() 4220{ 4221 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4222 sp<ThreadBase> thread = mThread.promote(); 4223 if (thread != 0) { 4224 Mutex::Autolock _l(thread->mLock); 4225 if (mState == ACTIVE || mState == RESUMING) { 4226 mState = PAUSING; 4227 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4228 if (!isOutputTrack()) { 4229 thread->mLock.unlock(); 4230 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4231 thread->mLock.lock(); 4232 4233#ifdef ADD_BATTERY_DATA 4234 // to track the speaker usage 4235 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4236#endif 4237 } 4238 } 4239 } 4240} 4241 4242void AudioFlinger::PlaybackThread::Track::flush() 4243{ 4244 ALOGV("flush(%d)", mName); 4245 sp<ThreadBase> thread = mThread.promote(); 4246 if (thread != 0) { 4247 Mutex::Autolock _l(thread->mLock); 4248 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 4249 return; 4250 } 4251 // No point remaining in PAUSED state after a flush => go to 4252 // STOPPED state 4253 mState = STOPPED; 4254 4255 // do not reset the track if it is still in the process of being stopped or paused. 4256 // this will be done by prepareTracks_l() when the track is stopped. 4257 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4258 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4259 reset(); 4260 } 4261 } 4262} 4263 4264void AudioFlinger::PlaybackThread::Track::reset() 4265{ 4266 // Do not reset twice to avoid discarding data written just after a flush and before 4267 // the audioflinger thread detects the track is stopped. 4268 if (!mResetDone) { 4269 TrackBase::reset(); 4270 // Force underrun condition to avoid false underrun callback until first data is 4271 // written to buffer 4272 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4273 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4274 mFillingUpStatus = FS_FILLING; 4275 mResetDone = true; 4276 mPresentationCompleteFrames = 0; 4277 } 4278} 4279 4280void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4281{ 4282 mMute = muted; 4283} 4284 4285status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4286{ 4287 status_t status = DEAD_OBJECT; 4288 sp<ThreadBase> thread = mThread.promote(); 4289 if (thread != 0) { 4290 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4291 status = playbackThread->attachAuxEffect(this, EffectId); 4292 } 4293 return status; 4294} 4295 4296void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4297{ 4298 mAuxEffectId = EffectId; 4299 mAuxBuffer = buffer; 4300} 4301 4302bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4303 size_t audioHalFrames) 4304{ 4305 // a track is considered presented when the total number of frames written to audio HAL 4306 // corresponds to the number of frames written when presentationComplete() is called for the 4307 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4308 if (mPresentationCompleteFrames == 0) { 4309 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4310 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4311 mPresentationCompleteFrames, audioHalFrames); 4312 } 4313 if (framesWritten >= mPresentationCompleteFrames) { 4314 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4315 mSessionId, framesWritten); 4316 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4317 mPresentationCompleteFrames = 0; 4318 return true; 4319 } 4320 return false; 4321} 4322 4323void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4324{ 4325 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4326 if (mSyncEvents[i]->type() == type) { 4327 mSyncEvents[i]->trigger(); 4328 mSyncEvents.removeAt(i); 4329 i--; 4330 } 4331 } 4332} 4333 4334// implement VolumeBufferProvider interface 4335 4336uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4337{ 4338 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4339 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4340 uint32_t vlr = mCblk->getVolumeLR(); 4341 uint32_t vl = vlr & 0xFFFF; 4342 uint32_t vr = vlr >> 16; 4343 // track volumes come from shared memory, so can't be trusted and must be clamped 4344 if (vl > MAX_GAIN_INT) { 4345 vl = MAX_GAIN_INT; 4346 } 4347 if (vr > MAX_GAIN_INT) { 4348 vr = MAX_GAIN_INT; 4349 } 4350 // now apply the cached master volume and stream type volume; 4351 // this is trusted but lacks any synchronization or barrier so may be stale 4352 float v = mCachedVolume; 4353 vl *= v; 4354 vr *= v; 4355 // re-combine into U4.16 4356 vlr = (vr << 16) | (vl & 0xFFFF); 4357 // FIXME look at mute, pause, and stop flags 4358 return vlr; 4359} 4360 4361// timed audio tracks 4362 4363sp<AudioFlinger::PlaybackThread::TimedTrack> 4364AudioFlinger::PlaybackThread::TimedTrack::create( 4365 PlaybackThread *thread, 4366 const sp<Client>& client, 4367 audio_stream_type_t streamType, 4368 uint32_t sampleRate, 4369 audio_format_t format, 4370 uint32_t channelMask, 4371 int frameCount, 4372 const sp<IMemory>& sharedBuffer, 4373 int sessionId) { 4374 if (!client->reserveTimedTrack()) 4375 return NULL; 4376 4377 return new TimedTrack( 4378 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4379 sharedBuffer, sessionId); 4380} 4381 4382AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4383 PlaybackThread *thread, 4384 const sp<Client>& client, 4385 audio_stream_type_t streamType, 4386 uint32_t sampleRate, 4387 audio_format_t format, 4388 uint32_t channelMask, 4389 int frameCount, 4390 const sp<IMemory>& sharedBuffer, 4391 int sessionId) 4392 : Track(thread, client, streamType, sampleRate, format, channelMask, 4393 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4394 mQueueHeadInFlight(false), 4395 mTrimQueueHeadOnRelease(false), 4396 mFramesPendingInQueue(0), 4397 mTimedSilenceBuffer(NULL), 4398 mTimedSilenceBufferSize(0), 4399 mTimedAudioOutputOnTime(false), 4400 mMediaTimeTransformValid(false) 4401{ 4402 LocalClock lc; 4403 mLocalTimeFreq = lc.getLocalFreq(); 4404 4405 mLocalTimeToSampleTransform.a_zero = 0; 4406 mLocalTimeToSampleTransform.b_zero = 0; 4407 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4408 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4409 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4410 &mLocalTimeToSampleTransform.a_to_b_denom); 4411 4412 mMediaTimeToSampleTransform.a_zero = 0; 4413 mMediaTimeToSampleTransform.b_zero = 0; 4414 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4415 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4416 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4417 &mMediaTimeToSampleTransform.a_to_b_denom); 4418} 4419 4420AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4421 mClient->releaseTimedTrack(); 4422 delete [] mTimedSilenceBuffer; 4423} 4424 4425status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4426 size_t size, sp<IMemory>* buffer) { 4427 4428 Mutex::Autolock _l(mTimedBufferQueueLock); 4429 4430 trimTimedBufferQueue_l(); 4431 4432 // lazily initialize the shared memory heap for timed buffers 4433 if (mTimedMemoryDealer == NULL) { 4434 const int kTimedBufferHeapSize = 512 << 10; 4435 4436 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4437 "AudioFlingerTimed"); 4438 if (mTimedMemoryDealer == NULL) 4439 return NO_MEMORY; 4440 } 4441 4442 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4443 if (newBuffer == NULL) { 4444 newBuffer = mTimedMemoryDealer->allocate(size); 4445 if (newBuffer == NULL) 4446 return NO_MEMORY; 4447 } 4448 4449 *buffer = newBuffer; 4450 return NO_ERROR; 4451} 4452 4453// caller must hold mTimedBufferQueueLock 4454void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4455 int64_t mediaTimeNow; 4456 { 4457 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4458 if (!mMediaTimeTransformValid) 4459 return; 4460 4461 int64_t targetTimeNow; 4462 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4463 ? mCCHelper.getCommonTime(&targetTimeNow) 4464 : mCCHelper.getLocalTime(&targetTimeNow); 4465 4466 if (OK != res) 4467 return; 4468 4469 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4470 &mediaTimeNow)) { 4471 return; 4472 } 4473 } 4474 4475 size_t trimEnd; 4476 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4477 int64_t bufEnd; 4478 4479 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4480 // We have a next buffer. Just use its PTS as the PTS of the frame 4481 // following the last frame in this buffer. If the stream is sparse 4482 // (ie, there are deliberate gaps left in the stream which should be 4483 // filled with silence by the TimedAudioTrack), then this can result 4484 // in one extra buffer being left un-trimmed when it could have 4485 // been. In general, this is not typical, and we would rather 4486 // optimized away the TS calculation below for the more common case 4487 // where PTSes are contiguous. 4488 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4489 } else { 4490 // We have no next buffer. Compute the PTS of the frame following 4491 // the last frame in this buffer by computing the duration of of 4492 // this frame in media time units and adding it to the PTS of the 4493 // buffer. 4494 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4495 / mCblk->frameSize; 4496 4497 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4498 &bufEnd)) { 4499 ALOGE("Failed to convert frame count of %lld to media time" 4500 " duration" " (scale factor %d/%u) in %s", 4501 frameCount, 4502 mMediaTimeToSampleTransform.a_to_b_numer, 4503 mMediaTimeToSampleTransform.a_to_b_denom, 4504 __PRETTY_FUNCTION__); 4505 break; 4506 } 4507 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4508 } 4509 4510 if (bufEnd > mediaTimeNow) 4511 break; 4512 4513 // Is the buffer we want to use in the middle of a mix operation right 4514 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4515 // from the mixer which should be coming back shortly. 4516 if (!trimEnd && mQueueHeadInFlight) { 4517 mTrimQueueHeadOnRelease = true; 4518 } 4519 } 4520 4521 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4522 if (trimStart < trimEnd) { 4523 // Update the bookkeeping for framesReady() 4524 for (size_t i = trimStart; i < trimEnd; ++i) { 4525 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4526 } 4527 4528 // Now actually remove the buffers from the queue. 4529 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4530 } 4531} 4532 4533void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4534 const char* logTag) { 4535 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4536 "%s called (reason \"%s\"), but timed buffer queue has no" 4537 " elements to trim.", __FUNCTION__, logTag); 4538 4539 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4540 mTimedBufferQueue.removeAt(0); 4541} 4542 4543void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4544 const TimedBuffer& buf, 4545 const char* logTag) { 4546 uint32_t bufBytes = buf.buffer()->size(); 4547 uint32_t consumedAlready = buf.position(); 4548 4549 ALOG_ASSERT(consumedAlready <= bufBytes, 4550 "Bad bookkeeping while updating frames pending. Timed buffer is" 4551 " only %u bytes long, but claims to have consumed %u" 4552 " bytes. (update reason: \"%s\")", 4553 bufBytes, consumedAlready, logTag); 4554 4555 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4556 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4557 "Bad bookkeeping while updating frames pending. Should have at" 4558 " least %u queued frames, but we think we have only %u. (update" 4559 " reason: \"%s\")", 4560 bufFrames, mFramesPendingInQueue, logTag); 4561 4562 mFramesPendingInQueue -= bufFrames; 4563} 4564 4565status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4566 const sp<IMemory>& buffer, int64_t pts) { 4567 4568 { 4569 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4570 if (!mMediaTimeTransformValid) 4571 return INVALID_OPERATION; 4572 } 4573 4574 Mutex::Autolock _l(mTimedBufferQueueLock); 4575 4576 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4577 mFramesPendingInQueue += bufFrames; 4578 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4579 4580 return NO_ERROR; 4581} 4582 4583status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4584 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4585 4586 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4587 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4588 target); 4589 4590 if (!(target == TimedAudioTrack::LOCAL_TIME || 4591 target == TimedAudioTrack::COMMON_TIME)) { 4592 return BAD_VALUE; 4593 } 4594 4595 Mutex::Autolock lock(mMediaTimeTransformLock); 4596 mMediaTimeTransform = xform; 4597 mMediaTimeTransformTarget = target; 4598 mMediaTimeTransformValid = true; 4599 4600 return NO_ERROR; 4601} 4602 4603#define min(a, b) ((a) < (b) ? (a) : (b)) 4604 4605// implementation of getNextBuffer for tracks whose buffers have timestamps 4606status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4607 AudioBufferProvider::Buffer* buffer, int64_t pts) 4608{ 4609 if (pts == AudioBufferProvider::kInvalidPTS) { 4610 buffer->raw = 0; 4611 buffer->frameCount = 0; 4612 mTimedAudioOutputOnTime = false; 4613 return INVALID_OPERATION; 4614 } 4615 4616 Mutex::Autolock _l(mTimedBufferQueueLock); 4617 4618 ALOG_ASSERT(!mQueueHeadInFlight, 4619 "getNextBuffer called without releaseBuffer!"); 4620 4621 while (true) { 4622 4623 // if we have no timed buffers, then fail 4624 if (mTimedBufferQueue.isEmpty()) { 4625 buffer->raw = 0; 4626 buffer->frameCount = 0; 4627 return NOT_ENOUGH_DATA; 4628 } 4629 4630 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4631 4632 // calculate the PTS of the head of the timed buffer queue expressed in 4633 // local time 4634 int64_t headLocalPTS; 4635 { 4636 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4637 4638 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4639 4640 if (mMediaTimeTransform.a_to_b_denom == 0) { 4641 // the transform represents a pause, so yield silence 4642 timedYieldSilence_l(buffer->frameCount, buffer); 4643 return NO_ERROR; 4644 } 4645 4646 int64_t transformedPTS; 4647 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4648 &transformedPTS)) { 4649 // the transform failed. this shouldn't happen, but if it does 4650 // then just drop this buffer 4651 ALOGW("timedGetNextBuffer transform failed"); 4652 buffer->raw = 0; 4653 buffer->frameCount = 0; 4654 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4655 return NO_ERROR; 4656 } 4657 4658 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4659 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4660 &headLocalPTS)) { 4661 buffer->raw = 0; 4662 buffer->frameCount = 0; 4663 return INVALID_OPERATION; 4664 } 4665 } else { 4666 headLocalPTS = transformedPTS; 4667 } 4668 } 4669 4670 // adjust the head buffer's PTS to reflect the portion of the head buffer 4671 // that has already been consumed 4672 int64_t effectivePTS = headLocalPTS + 4673 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4674 4675 // Calculate the delta in samples between the head of the input buffer 4676 // queue and the start of the next output buffer that will be written. 4677 // If the transformation fails because of over or underflow, it means 4678 // that the sample's position in the output stream is so far out of 4679 // whack that it should just be dropped. 4680 int64_t sampleDelta; 4681 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4682 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4683 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4684 " mix"); 4685 continue; 4686 } 4687 if (!mLocalTimeToSampleTransform.doForwardTransform( 4688 (effectivePTS - pts) << 32, &sampleDelta)) { 4689 ALOGV("*** too late during sample rate transform: dropped buffer"); 4690 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 4691 continue; 4692 } 4693 4694 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 4695 " sampleDelta=[%d.%08x]", 4696 head.pts(), head.position(), pts, 4697 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 4698 + (sampleDelta >> 32)), 4699 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4700 4701 // if the delta between the ideal placement for the next input sample and 4702 // the current output position is within this threshold, then we will 4703 // concatenate the next input samples to the previous output 4704 const int64_t kSampleContinuityThreshold = 4705 (static_cast<int64_t>(sampleRate()) << 32) / 250; 4706 4707 // if this is the first buffer of audio that we're emitting from this track 4708 // then it should be almost exactly on time. 4709 const int64_t kSampleStartupThreshold = 1LL << 32; 4710 4711 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4712 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4713 // the next input is close enough to being on time, so concatenate it 4714 // with the last output 4715 timedYieldSamples_l(buffer); 4716 4717 ALOGVV("*** on time: head.pos=%d frameCount=%u", 4718 head.position(), buffer->frameCount); 4719 return NO_ERROR; 4720 } 4721 4722 // Looks like our output is not on time. Reset our on timed status. 4723 // Next time we mix samples from our input queue, then should be within 4724 // the StartupThreshold. 4725 mTimedAudioOutputOnTime = false; 4726 if (sampleDelta > 0) { 4727 // the gap between the current output position and the proper start of 4728 // the next input sample is too big, so fill it with silence 4729 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4730 4731 timedYieldSilence_l(framesUntilNextInput, buffer); 4732 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4733 return NO_ERROR; 4734 } else { 4735 // the next input sample is late 4736 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4737 size_t onTimeSamplePosition = 4738 head.position() + lateFrames * mCblk->frameSize; 4739 4740 if (onTimeSamplePosition > head.buffer()->size()) { 4741 // all the remaining samples in the head are too late, so 4742 // drop it and move on 4743 ALOGV("*** too late: dropped buffer"); 4744 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 4745 continue; 4746 } else { 4747 // skip over the late samples 4748 head.setPosition(onTimeSamplePosition); 4749 4750 // yield the available samples 4751 timedYieldSamples_l(buffer); 4752 4753 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4754 return NO_ERROR; 4755 } 4756 } 4757 } 4758} 4759 4760// Yield samples from the timed buffer queue head up to the given output 4761// buffer's capacity. 4762// 4763// Caller must hold mTimedBufferQueueLock 4764void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 4765 AudioBufferProvider::Buffer* buffer) { 4766 4767 const TimedBuffer& head = mTimedBufferQueue[0]; 4768 4769 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4770 head.position()); 4771 4772 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4773 mCblk->frameSize); 4774 size_t framesRequested = buffer->frameCount; 4775 buffer->frameCount = min(framesLeftInHead, framesRequested); 4776 4777 mQueueHeadInFlight = true; 4778 mTimedAudioOutputOnTime = true; 4779} 4780 4781// Yield samples of silence up to the given output buffer's capacity 4782// 4783// Caller must hold mTimedBufferQueueLock 4784void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 4785 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4786 4787 // lazily allocate a buffer filled with silence 4788 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4789 delete [] mTimedSilenceBuffer; 4790 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4791 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4792 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4793 } 4794 4795 buffer->raw = mTimedSilenceBuffer; 4796 size_t framesRequested = buffer->frameCount; 4797 buffer->frameCount = min(numFrames, framesRequested); 4798 4799 mTimedAudioOutputOnTime = false; 4800} 4801 4802// AudioBufferProvider interface 4803void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4804 AudioBufferProvider::Buffer* buffer) { 4805 4806 Mutex::Autolock _l(mTimedBufferQueueLock); 4807 4808 // If the buffer which was just released is part of the buffer at the head 4809 // of the queue, be sure to update the amt of the buffer which has been 4810 // consumed. If the buffer being returned is not part of the head of the 4811 // queue, its either because the buffer is part of the silence buffer, or 4812 // because the head of the timed queue was trimmed after the mixer called 4813 // getNextBuffer but before the mixer called releaseBuffer. 4814 if (buffer->raw == mTimedSilenceBuffer) { 4815 ALOG_ASSERT(!mQueueHeadInFlight, 4816 "Queue head in flight during release of silence buffer!"); 4817 goto done; 4818 } 4819 4820 ALOG_ASSERT(mQueueHeadInFlight, 4821 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 4822 " head in flight."); 4823 4824 if (mTimedBufferQueue.size()) { 4825 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4826 4827 void* start = head.buffer()->pointer(); 4828 void* end = reinterpret_cast<void*>( 4829 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 4830 + head.buffer()->size()); 4831 4832 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 4833 "released buffer not within the head of the timed buffer" 4834 " queue; qHead = [%p, %p], released buffer = %p", 4835 start, end, buffer->raw); 4836 4837 head.setPosition(head.position() + 4838 (buffer->frameCount * mCblk->frameSize)); 4839 mQueueHeadInFlight = false; 4840 4841 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 4842 "Bad bookkeeping during releaseBuffer! Should have at" 4843 " least %u queued frames, but we think we have only %u", 4844 buffer->frameCount, mFramesPendingInQueue); 4845 4846 mFramesPendingInQueue -= buffer->frameCount; 4847 4848 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 4849 || mTrimQueueHeadOnRelease) { 4850 trimTimedBufferQueueHead_l("releaseBuffer"); 4851 mTrimQueueHeadOnRelease = false; 4852 } 4853 } else { 4854 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 4855 " buffers in the timed buffer queue"); 4856 } 4857 4858done: 4859 buffer->raw = 0; 4860 buffer->frameCount = 0; 4861} 4862 4863uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4864 Mutex::Autolock _l(mTimedBufferQueueLock); 4865 return mFramesPendingInQueue; 4866} 4867 4868AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4869 : mPTS(0), mPosition(0) {} 4870 4871AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4872 const sp<IMemory>& buffer, int64_t pts) 4873 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4874 4875// ---------------------------------------------------------------------------- 4876 4877// RecordTrack constructor must be called with AudioFlinger::mLock held 4878AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4879 RecordThread *thread, 4880 const sp<Client>& client, 4881 uint32_t sampleRate, 4882 audio_format_t format, 4883 uint32_t channelMask, 4884 int frameCount, 4885 int sessionId) 4886 : TrackBase(thread, client, sampleRate, format, 4887 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4888 mOverflow(false) 4889{ 4890 if (mCblk != NULL) { 4891 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4892 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4893 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4894 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4895 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4896 } else { 4897 mCblk->frameSize = sizeof(int8_t); 4898 } 4899 } 4900} 4901 4902AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4903{ 4904 sp<ThreadBase> thread = mThread.promote(); 4905 if (thread != 0) { 4906 AudioSystem::releaseInput(thread->id()); 4907 } 4908} 4909 4910// AudioBufferProvider interface 4911status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4912{ 4913 audio_track_cblk_t* cblk = this->cblk(); 4914 uint32_t framesAvail; 4915 uint32_t framesReq = buffer->frameCount; 4916 4917 // Check if last stepServer failed, try to step now 4918 if (mStepServerFailed) { 4919 if (!step()) goto getNextBuffer_exit; 4920 ALOGV("stepServer recovered"); 4921 mStepServerFailed = false; 4922 } 4923 4924 framesAvail = cblk->framesAvailable_l(); 4925 4926 if (CC_LIKELY(framesAvail)) { 4927 uint32_t s = cblk->server; 4928 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4929 4930 if (framesReq > framesAvail) { 4931 framesReq = framesAvail; 4932 } 4933 if (framesReq > bufferEnd - s) { 4934 framesReq = bufferEnd - s; 4935 } 4936 4937 buffer->raw = getBuffer(s, framesReq); 4938 if (buffer->raw == NULL) goto getNextBuffer_exit; 4939 4940 buffer->frameCount = framesReq; 4941 return NO_ERROR; 4942 } 4943 4944getNextBuffer_exit: 4945 buffer->raw = NULL; 4946 buffer->frameCount = 0; 4947 return NOT_ENOUGH_DATA; 4948} 4949 4950status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 4951 int triggerSession) 4952{ 4953 sp<ThreadBase> thread = mThread.promote(); 4954 if (thread != 0) { 4955 RecordThread *recordThread = (RecordThread *)thread.get(); 4956 return recordThread->start(this, event, triggerSession); 4957 } else { 4958 return BAD_VALUE; 4959 } 4960} 4961 4962void AudioFlinger::RecordThread::RecordTrack::stop() 4963{ 4964 sp<ThreadBase> thread = mThread.promote(); 4965 if (thread != 0) { 4966 RecordThread *recordThread = (RecordThread *)thread.get(); 4967 recordThread->stop(this); 4968 TrackBase::reset(); 4969 // Force overrun condition to avoid false overrun callback until first data is 4970 // read from buffer 4971 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4972 } 4973} 4974 4975void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4976{ 4977 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4978 (mClient == 0) ? getpid_cached : mClient->pid(), 4979 mFormat, 4980 mChannelMask, 4981 mSessionId, 4982 mFrameCount, 4983 mState, 4984 mCblk->sampleRate, 4985 mCblk->server, 4986 mCblk->user); 4987} 4988 4989 4990// ---------------------------------------------------------------------------- 4991 4992AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4993 PlaybackThread *playbackThread, 4994 DuplicatingThread *sourceThread, 4995 uint32_t sampleRate, 4996 audio_format_t format, 4997 uint32_t channelMask, 4998 int frameCount) 4999 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5000 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5001 mActive(false), mSourceThread(sourceThread) 5002{ 5003 5004 if (mCblk != NULL) { 5005 mCblk->flags |= CBLK_DIRECTION_OUT; 5006 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5007 mOutBuffer.frameCount = 0; 5008 playbackThread->mTracks.add(this); 5009 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5010 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5011 mCblk, mBuffer, mCblk->buffers, 5012 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5013 } else { 5014 ALOGW("Error creating output track on thread %p", playbackThread); 5015 } 5016} 5017 5018AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5019{ 5020 clearBufferQueue(); 5021} 5022 5023status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5024 int triggerSession) 5025{ 5026 status_t status = Track::start(event, triggerSession); 5027 if (status != NO_ERROR) { 5028 return status; 5029 } 5030 5031 mActive = true; 5032 mRetryCount = 127; 5033 return status; 5034} 5035 5036void AudioFlinger::PlaybackThread::OutputTrack::stop() 5037{ 5038 Track::stop(); 5039 clearBufferQueue(); 5040 mOutBuffer.frameCount = 0; 5041 mActive = false; 5042} 5043 5044bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5045{ 5046 Buffer *pInBuffer; 5047 Buffer inBuffer; 5048 uint32_t channelCount = mChannelCount; 5049 bool outputBufferFull = false; 5050 inBuffer.frameCount = frames; 5051 inBuffer.i16 = data; 5052 5053 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5054 5055 if (!mActive && frames != 0) { 5056 start(); 5057 sp<ThreadBase> thread = mThread.promote(); 5058 if (thread != 0) { 5059 MixerThread *mixerThread = (MixerThread *)thread.get(); 5060 if (mCblk->frameCount > frames){ 5061 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5062 uint32_t startFrames = (mCblk->frameCount - frames); 5063 pInBuffer = new Buffer; 5064 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5065 pInBuffer->frameCount = startFrames; 5066 pInBuffer->i16 = pInBuffer->mBuffer; 5067 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5068 mBufferQueue.add(pInBuffer); 5069 } else { 5070 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5071 } 5072 } 5073 } 5074 } 5075 5076 while (waitTimeLeftMs) { 5077 // First write pending buffers, then new data 5078 if (mBufferQueue.size()) { 5079 pInBuffer = mBufferQueue.itemAt(0); 5080 } else { 5081 pInBuffer = &inBuffer; 5082 } 5083 5084 if (pInBuffer->frameCount == 0) { 5085 break; 5086 } 5087 5088 if (mOutBuffer.frameCount == 0) { 5089 mOutBuffer.frameCount = pInBuffer->frameCount; 5090 nsecs_t startTime = systemTime(); 5091 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5092 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5093 outputBufferFull = true; 5094 break; 5095 } 5096 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5097 if (waitTimeLeftMs >= waitTimeMs) { 5098 waitTimeLeftMs -= waitTimeMs; 5099 } else { 5100 waitTimeLeftMs = 0; 5101 } 5102 } 5103 5104 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5105 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5106 mCblk->stepUser(outFrames); 5107 pInBuffer->frameCount -= outFrames; 5108 pInBuffer->i16 += outFrames * channelCount; 5109 mOutBuffer.frameCount -= outFrames; 5110 mOutBuffer.i16 += outFrames * channelCount; 5111 5112 if (pInBuffer->frameCount == 0) { 5113 if (mBufferQueue.size()) { 5114 mBufferQueue.removeAt(0); 5115 delete [] pInBuffer->mBuffer; 5116 delete pInBuffer; 5117 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5118 } else { 5119 break; 5120 } 5121 } 5122 } 5123 5124 // If we could not write all frames, allocate a buffer and queue it for next time. 5125 if (inBuffer.frameCount) { 5126 sp<ThreadBase> thread = mThread.promote(); 5127 if (thread != 0 && !thread->standby()) { 5128 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5129 pInBuffer = new Buffer; 5130 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5131 pInBuffer->frameCount = inBuffer.frameCount; 5132 pInBuffer->i16 = pInBuffer->mBuffer; 5133 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5134 mBufferQueue.add(pInBuffer); 5135 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5136 } else { 5137 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5138 } 5139 } 5140 } 5141 5142 // Calling write() with a 0 length buffer, means that no more data will be written: 5143 // If no more buffers are pending, fill output track buffer to make sure it is started 5144 // by output mixer. 5145 if (frames == 0 && mBufferQueue.size() == 0) { 5146 if (mCblk->user < mCblk->frameCount) { 5147 frames = mCblk->frameCount - mCblk->user; 5148 pInBuffer = new Buffer; 5149 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5150 pInBuffer->frameCount = frames; 5151 pInBuffer->i16 = pInBuffer->mBuffer; 5152 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5153 mBufferQueue.add(pInBuffer); 5154 } else if (mActive) { 5155 stop(); 5156 } 5157 } 5158 5159 return outputBufferFull; 5160} 5161 5162status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5163{ 5164 int active; 5165 status_t result; 5166 audio_track_cblk_t* cblk = mCblk; 5167 uint32_t framesReq = buffer->frameCount; 5168 5169// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5170 buffer->frameCount = 0; 5171 5172 uint32_t framesAvail = cblk->framesAvailable(); 5173 5174 5175 if (framesAvail == 0) { 5176 Mutex::Autolock _l(cblk->lock); 5177 goto start_loop_here; 5178 while (framesAvail == 0) { 5179 active = mActive; 5180 if (CC_UNLIKELY(!active)) { 5181 ALOGV("Not active and NO_MORE_BUFFERS"); 5182 return NO_MORE_BUFFERS; 5183 } 5184 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5185 if (result != NO_ERROR) { 5186 return NO_MORE_BUFFERS; 5187 } 5188 // read the server count again 5189 start_loop_here: 5190 framesAvail = cblk->framesAvailable_l(); 5191 } 5192 } 5193 5194// if (framesAvail < framesReq) { 5195// return NO_MORE_BUFFERS; 5196// } 5197 5198 if (framesReq > framesAvail) { 5199 framesReq = framesAvail; 5200 } 5201 5202 uint32_t u = cblk->user; 5203 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5204 5205 if (framesReq > bufferEnd - u) { 5206 framesReq = bufferEnd - u; 5207 } 5208 5209 buffer->frameCount = framesReq; 5210 buffer->raw = (void *)cblk->buffer(u); 5211 return NO_ERROR; 5212} 5213 5214 5215void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5216{ 5217 size_t size = mBufferQueue.size(); 5218 5219 for (size_t i = 0; i < size; i++) { 5220 Buffer *pBuffer = mBufferQueue.itemAt(i); 5221 delete [] pBuffer->mBuffer; 5222 delete pBuffer; 5223 } 5224 mBufferQueue.clear(); 5225} 5226 5227// ---------------------------------------------------------------------------- 5228 5229AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5230 : RefBase(), 5231 mAudioFlinger(audioFlinger), 5232 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5233 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5234 mPid(pid), 5235 mTimedTrackCount(0) 5236{ 5237 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5238} 5239 5240// Client destructor must be called with AudioFlinger::mLock held 5241AudioFlinger::Client::~Client() 5242{ 5243 mAudioFlinger->removeClient_l(mPid); 5244} 5245 5246sp<MemoryDealer> AudioFlinger::Client::heap() const 5247{ 5248 return mMemoryDealer; 5249} 5250 5251// Reserve one of the limited slots for a timed audio track associated 5252// with this client 5253bool AudioFlinger::Client::reserveTimedTrack() 5254{ 5255 const int kMaxTimedTracksPerClient = 4; 5256 5257 Mutex::Autolock _l(mTimedTrackLock); 5258 5259 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5260 ALOGW("can not create timed track - pid %d has exceeded the limit", 5261 mPid); 5262 return false; 5263 } 5264 5265 mTimedTrackCount++; 5266 return true; 5267} 5268 5269// Release a slot for a timed audio track 5270void AudioFlinger::Client::releaseTimedTrack() 5271{ 5272 Mutex::Autolock _l(mTimedTrackLock); 5273 mTimedTrackCount--; 5274} 5275 5276// ---------------------------------------------------------------------------- 5277 5278AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5279 const sp<IAudioFlingerClient>& client, 5280 pid_t pid) 5281 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5282{ 5283} 5284 5285AudioFlinger::NotificationClient::~NotificationClient() 5286{ 5287} 5288 5289void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5290{ 5291 sp<NotificationClient> keep(this); 5292 mAudioFlinger->removeNotificationClient(mPid); 5293} 5294 5295// ---------------------------------------------------------------------------- 5296 5297AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5298 : BnAudioTrack(), 5299 mTrack(track) 5300{ 5301} 5302 5303AudioFlinger::TrackHandle::~TrackHandle() { 5304 // just stop the track on deletion, associated resources 5305 // will be freed from the main thread once all pending buffers have 5306 // been played. Unless it's not in the active track list, in which 5307 // case we free everything now... 5308 mTrack->destroy(); 5309} 5310 5311sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5312 return mTrack->getCblk(); 5313} 5314 5315status_t AudioFlinger::TrackHandle::start() { 5316 return mTrack->start(); 5317} 5318 5319void AudioFlinger::TrackHandle::stop() { 5320 mTrack->stop(); 5321} 5322 5323void AudioFlinger::TrackHandle::flush() { 5324 mTrack->flush(); 5325} 5326 5327void AudioFlinger::TrackHandle::mute(bool e) { 5328 mTrack->mute(e); 5329} 5330 5331void AudioFlinger::TrackHandle::pause() { 5332 mTrack->pause(); 5333} 5334 5335status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5336{ 5337 return mTrack->attachAuxEffect(EffectId); 5338} 5339 5340status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5341 sp<IMemory>* buffer) { 5342 if (!mTrack->isTimedTrack()) 5343 return INVALID_OPERATION; 5344 5345 PlaybackThread::TimedTrack* tt = 5346 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5347 return tt->allocateTimedBuffer(size, buffer); 5348} 5349 5350status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5351 int64_t pts) { 5352 if (!mTrack->isTimedTrack()) 5353 return INVALID_OPERATION; 5354 5355 PlaybackThread::TimedTrack* tt = 5356 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5357 return tt->queueTimedBuffer(buffer, pts); 5358} 5359 5360status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5361 const LinearTransform& xform, int target) { 5362 5363 if (!mTrack->isTimedTrack()) 5364 return INVALID_OPERATION; 5365 5366 PlaybackThread::TimedTrack* tt = 5367 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5368 return tt->setMediaTimeTransform( 5369 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5370} 5371 5372status_t AudioFlinger::TrackHandle::onTransact( 5373 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5374{ 5375 return BnAudioTrack::onTransact(code, data, reply, flags); 5376} 5377 5378// ---------------------------------------------------------------------------- 5379 5380sp<IAudioRecord> AudioFlinger::openRecord( 5381 pid_t pid, 5382 audio_io_handle_t input, 5383 uint32_t sampleRate, 5384 audio_format_t format, 5385 uint32_t channelMask, 5386 int frameCount, 5387 IAudioFlinger::track_flags_t flags, 5388 int *sessionId, 5389 status_t *status) 5390{ 5391 sp<RecordThread::RecordTrack> recordTrack; 5392 sp<RecordHandle> recordHandle; 5393 sp<Client> client; 5394 status_t lStatus; 5395 RecordThread *thread; 5396 size_t inFrameCount; 5397 int lSessionId; 5398 5399 // check calling permissions 5400 if (!recordingAllowed()) { 5401 lStatus = PERMISSION_DENIED; 5402 goto Exit; 5403 } 5404 5405 // add client to list 5406 { // scope for mLock 5407 Mutex::Autolock _l(mLock); 5408 thread = checkRecordThread_l(input); 5409 if (thread == NULL) { 5410 lStatus = BAD_VALUE; 5411 goto Exit; 5412 } 5413 5414 client = registerPid_l(pid); 5415 5416 // If no audio session id is provided, create one here 5417 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5418 lSessionId = *sessionId; 5419 } else { 5420 lSessionId = nextUniqueId(); 5421 if (sessionId != NULL) { 5422 *sessionId = lSessionId; 5423 } 5424 } 5425 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5426 recordTrack = thread->createRecordTrack_l(client, 5427 sampleRate, 5428 format, 5429 channelMask, 5430 frameCount, 5431 lSessionId, 5432 &lStatus); 5433 } 5434 if (lStatus != NO_ERROR) { 5435 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5436 // destructor is called by the TrackBase destructor with mLock held 5437 client.clear(); 5438 recordTrack.clear(); 5439 goto Exit; 5440 } 5441 5442 // return to handle to client 5443 recordHandle = new RecordHandle(recordTrack); 5444 lStatus = NO_ERROR; 5445 5446Exit: 5447 if (status) { 5448 *status = lStatus; 5449 } 5450 return recordHandle; 5451} 5452 5453// ---------------------------------------------------------------------------- 5454 5455AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5456 : BnAudioRecord(), 5457 mRecordTrack(recordTrack) 5458{ 5459} 5460 5461AudioFlinger::RecordHandle::~RecordHandle() { 5462 stop(); 5463} 5464 5465sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5466 return mRecordTrack->getCblk(); 5467} 5468 5469status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5470 ALOGV("RecordHandle::start()"); 5471 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5472} 5473 5474void AudioFlinger::RecordHandle::stop() { 5475 ALOGV("RecordHandle::stop()"); 5476 mRecordTrack->stop(); 5477} 5478 5479status_t AudioFlinger::RecordHandle::onTransact( 5480 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5481{ 5482 return BnAudioRecord::onTransact(code, data, reply, flags); 5483} 5484 5485// ---------------------------------------------------------------------------- 5486 5487AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5488 AudioStreamIn *input, 5489 uint32_t sampleRate, 5490 uint32_t channels, 5491 audio_io_handle_t id, 5492 uint32_t device) : 5493 ThreadBase(audioFlinger, id, device, RECORD), 5494 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5495 // mRsmpInIndex and mInputBytes set by readInputParameters() 5496 mReqChannelCount(popcount(channels)), 5497 mReqSampleRate(sampleRate) 5498 // mBytesRead is only meaningful while active, and so is cleared in start() 5499 // (but might be better to also clear here for dump?) 5500{ 5501 snprintf(mName, kNameLength, "AudioIn_%X", id); 5502 5503 readInputParameters(); 5504} 5505 5506 5507AudioFlinger::RecordThread::~RecordThread() 5508{ 5509 delete[] mRsmpInBuffer; 5510 delete mResampler; 5511 delete[] mRsmpOutBuffer; 5512} 5513 5514void AudioFlinger::RecordThread::onFirstRef() 5515{ 5516 run(mName, PRIORITY_URGENT_AUDIO); 5517} 5518 5519status_t AudioFlinger::RecordThread::readyToRun() 5520{ 5521 status_t status = initCheck(); 5522 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5523 return status; 5524} 5525 5526bool AudioFlinger::RecordThread::threadLoop() 5527{ 5528 AudioBufferProvider::Buffer buffer; 5529 sp<RecordTrack> activeTrack; 5530 Vector< sp<EffectChain> > effectChains; 5531 5532 nsecs_t lastWarning = 0; 5533 5534 acquireWakeLock(); 5535 5536 // start recording 5537 while (!exitPending()) { 5538 5539 processConfigEvents(); 5540 5541 { // scope for mLock 5542 Mutex::Autolock _l(mLock); 5543 checkForNewParameters_l(); 5544 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5545 if (!mStandby) { 5546 mInput->stream->common.standby(&mInput->stream->common); 5547 mStandby = true; 5548 } 5549 5550 if (exitPending()) break; 5551 5552 releaseWakeLock_l(); 5553 ALOGV("RecordThread: loop stopping"); 5554 // go to sleep 5555 mWaitWorkCV.wait(mLock); 5556 ALOGV("RecordThread: loop starting"); 5557 acquireWakeLock_l(); 5558 continue; 5559 } 5560 if (mActiveTrack != 0) { 5561 if (mActiveTrack->mState == TrackBase::PAUSING) { 5562 if (!mStandby) { 5563 mInput->stream->common.standby(&mInput->stream->common); 5564 mStandby = true; 5565 } 5566 mActiveTrack.clear(); 5567 mStartStopCond.broadcast(); 5568 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5569 if (mReqChannelCount != mActiveTrack->channelCount()) { 5570 mActiveTrack.clear(); 5571 mStartStopCond.broadcast(); 5572 } else if (mBytesRead != 0) { 5573 // record start succeeds only if first read from audio input 5574 // succeeds 5575 if (mBytesRead > 0) { 5576 mActiveTrack->mState = TrackBase::ACTIVE; 5577 } else { 5578 mActiveTrack.clear(); 5579 } 5580 mStartStopCond.broadcast(); 5581 } 5582 mStandby = false; 5583 } 5584 } 5585 lockEffectChains_l(effectChains); 5586 } 5587 5588 if (mActiveTrack != 0) { 5589 if (mActiveTrack->mState != TrackBase::ACTIVE && 5590 mActiveTrack->mState != TrackBase::RESUMING) { 5591 unlockEffectChains(effectChains); 5592 usleep(kRecordThreadSleepUs); 5593 continue; 5594 } 5595 for (size_t i = 0; i < effectChains.size(); i ++) { 5596 effectChains[i]->process_l(); 5597 } 5598 5599 buffer.frameCount = mFrameCount; 5600 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5601 size_t framesOut = buffer.frameCount; 5602 if (mResampler == NULL) { 5603 // no resampling 5604 while (framesOut) { 5605 size_t framesIn = mFrameCount - mRsmpInIndex; 5606 if (framesIn) { 5607 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5608 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5609 if (framesIn > framesOut) 5610 framesIn = framesOut; 5611 mRsmpInIndex += framesIn; 5612 framesOut -= framesIn; 5613 if ((int)mChannelCount == mReqChannelCount || 5614 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5615 memcpy(dst, src, framesIn * mFrameSize); 5616 } else { 5617 int16_t *src16 = (int16_t *)src; 5618 int16_t *dst16 = (int16_t *)dst; 5619 if (mChannelCount == 1) { 5620 while (framesIn--) { 5621 *dst16++ = *src16; 5622 *dst16++ = *src16++; 5623 } 5624 } else { 5625 while (framesIn--) { 5626 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5627 src16 += 2; 5628 } 5629 } 5630 } 5631 } 5632 if (framesOut && mFrameCount == mRsmpInIndex) { 5633 if (framesOut == mFrameCount && 5634 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5635 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5636 framesOut = 0; 5637 } else { 5638 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5639 mRsmpInIndex = 0; 5640 } 5641 if (mBytesRead < 0) { 5642 ALOGE("Error reading audio input"); 5643 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5644 // Force input into standby so that it tries to 5645 // recover at next read attempt 5646 mInput->stream->common.standby(&mInput->stream->common); 5647 usleep(kRecordThreadSleepUs); 5648 } 5649 mRsmpInIndex = mFrameCount; 5650 framesOut = 0; 5651 buffer.frameCount = 0; 5652 } 5653 } 5654 } 5655 } else { 5656 // resampling 5657 5658 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5659 // alter output frame count as if we were expecting stereo samples 5660 if (mChannelCount == 1 && mReqChannelCount == 1) { 5661 framesOut >>= 1; 5662 } 5663 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5664 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5665 // are 32 bit aligned which should be always true. 5666 if (mChannelCount == 2 && mReqChannelCount == 1) { 5667 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5668 // the resampler always outputs stereo samples: do post stereo to mono conversion 5669 int16_t *src = (int16_t *)mRsmpOutBuffer; 5670 int16_t *dst = buffer.i16; 5671 while (framesOut--) { 5672 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5673 src += 2; 5674 } 5675 } else { 5676 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5677 } 5678 5679 } 5680 if (mFramestoDrop == 0) { 5681 mActiveTrack->releaseBuffer(&buffer); 5682 } else { 5683 if (mFramestoDrop > 0) { 5684 mFramestoDrop -= buffer.frameCount; 5685 if (mFramestoDrop < 0) { 5686 mFramestoDrop = 0; 5687 } 5688 } 5689 } 5690 mActiveTrack->overflow(); 5691 } 5692 // client isn't retrieving buffers fast enough 5693 else { 5694 if (!mActiveTrack->setOverflow()) { 5695 nsecs_t now = systemTime(); 5696 if ((now - lastWarning) > kWarningThrottleNs) { 5697 ALOGW("RecordThread: buffer overflow"); 5698 lastWarning = now; 5699 } 5700 } 5701 // Release the processor for a while before asking for a new buffer. 5702 // This will give the application more chance to read from the buffer and 5703 // clear the overflow. 5704 usleep(kRecordThreadSleepUs); 5705 } 5706 } 5707 // enable changes in effect chain 5708 unlockEffectChains(effectChains); 5709 effectChains.clear(); 5710 } 5711 5712 if (!mStandby) { 5713 mInput->stream->common.standby(&mInput->stream->common); 5714 } 5715 mActiveTrack.clear(); 5716 5717 mStartStopCond.broadcast(); 5718 5719 releaseWakeLock(); 5720 5721 ALOGV("RecordThread %p exiting", this); 5722 return false; 5723} 5724 5725 5726sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5727 const sp<AudioFlinger::Client>& client, 5728 uint32_t sampleRate, 5729 audio_format_t format, 5730 int channelMask, 5731 int frameCount, 5732 int sessionId, 5733 status_t *status) 5734{ 5735 sp<RecordTrack> track; 5736 status_t lStatus; 5737 5738 lStatus = initCheck(); 5739 if (lStatus != NO_ERROR) { 5740 ALOGE("Audio driver not initialized."); 5741 goto Exit; 5742 } 5743 5744 { // scope for mLock 5745 Mutex::Autolock _l(mLock); 5746 5747 track = new RecordTrack(this, client, sampleRate, 5748 format, channelMask, frameCount, sessionId); 5749 5750 if (track->getCblk() == 0) { 5751 lStatus = NO_MEMORY; 5752 goto Exit; 5753 } 5754 5755 mTrack = track.get(); 5756 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5757 bool suspend = audio_is_bluetooth_sco_device( 5758 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5759 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5760 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5761 } 5762 lStatus = NO_ERROR; 5763 5764Exit: 5765 if (status) { 5766 *status = lStatus; 5767 } 5768 return track; 5769} 5770 5771status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5772 AudioSystem::sync_event_t event, 5773 int triggerSession) 5774{ 5775 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5776 sp<ThreadBase> strongMe = this; 5777 status_t status = NO_ERROR; 5778 5779 if (event == AudioSystem::SYNC_EVENT_NONE) { 5780 mSyncStartEvent.clear(); 5781 mFramestoDrop = 0; 5782 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5783 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5784 triggerSession, 5785 recordTrack->sessionId(), 5786 syncStartEventCallback, 5787 this); 5788 mFramestoDrop = -1; 5789 } 5790 5791 { 5792 AutoMutex lock(mLock); 5793 if (mActiveTrack != 0) { 5794 if (recordTrack != mActiveTrack.get()) { 5795 status = -EBUSY; 5796 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5797 mActiveTrack->mState = TrackBase::ACTIVE; 5798 } 5799 return status; 5800 } 5801 5802 recordTrack->mState = TrackBase::IDLE; 5803 mActiveTrack = recordTrack; 5804 mLock.unlock(); 5805 status_t status = AudioSystem::startInput(mId); 5806 mLock.lock(); 5807 if (status != NO_ERROR) { 5808 mActiveTrack.clear(); 5809 clearSyncStartEvent(); 5810 return status; 5811 } 5812 mRsmpInIndex = mFrameCount; 5813 mBytesRead = 0; 5814 if (mResampler != NULL) { 5815 mResampler->reset(); 5816 } 5817 mActiveTrack->mState = TrackBase::RESUMING; 5818 // signal thread to start 5819 ALOGV("Signal record thread"); 5820 mWaitWorkCV.signal(); 5821 // do not wait for mStartStopCond if exiting 5822 if (exitPending()) { 5823 mActiveTrack.clear(); 5824 status = INVALID_OPERATION; 5825 goto startError; 5826 } 5827 mStartStopCond.wait(mLock); 5828 if (mActiveTrack == 0) { 5829 ALOGV("Record failed to start"); 5830 status = BAD_VALUE; 5831 goto startError; 5832 } 5833 ALOGV("Record started OK"); 5834 return status; 5835 } 5836startError: 5837 AudioSystem::stopInput(mId); 5838 clearSyncStartEvent(); 5839 return status; 5840} 5841 5842void AudioFlinger::RecordThread::clearSyncStartEvent() 5843{ 5844 if (mSyncStartEvent != 0) { 5845 mSyncStartEvent->cancel(); 5846 } 5847 mSyncStartEvent.clear(); 5848} 5849 5850void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5851{ 5852 sp<SyncEvent> strongEvent = event.promote(); 5853 5854 if (strongEvent != 0) { 5855 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5856 me->handleSyncStartEvent(strongEvent); 5857 } 5858} 5859 5860void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5861{ 5862 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5863 mActiveTrack.get(), 5864 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5865 event->listenerSession()); 5866 5867 if (mActiveTrack != 0 && 5868 event == mSyncStartEvent) { 5869 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5870 // from audio HAL 5871 mFramestoDrop = mFrameCount * 2; 5872 mSyncStartEvent.clear(); 5873 } 5874} 5875 5876void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5877 ALOGV("RecordThread::stop"); 5878 sp<ThreadBase> strongMe = this; 5879 { 5880 AutoMutex lock(mLock); 5881 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5882 mActiveTrack->mState = TrackBase::PAUSING; 5883 // do not wait for mStartStopCond if exiting 5884 if (exitPending()) { 5885 return; 5886 } 5887 mStartStopCond.wait(mLock); 5888 // if we have been restarted, recordTrack == mActiveTrack.get() here 5889 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5890 mLock.unlock(); 5891 AudioSystem::stopInput(mId); 5892 mLock.lock(); 5893 ALOGV("Record stopped OK"); 5894 } 5895 } 5896 } 5897} 5898 5899bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5900{ 5901 return false; 5902} 5903 5904status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5905{ 5906 if (!isValidSyncEvent(event)) { 5907 return BAD_VALUE; 5908 } 5909 5910 Mutex::Autolock _l(mLock); 5911 5912 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5913 mTrack->setSyncEvent(event); 5914 return NO_ERROR; 5915 } 5916 return NAME_NOT_FOUND; 5917} 5918 5919status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5920{ 5921 const size_t SIZE = 256; 5922 char buffer[SIZE]; 5923 String8 result; 5924 5925 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5926 result.append(buffer); 5927 5928 if (mActiveTrack != 0) { 5929 result.append("Active Track:\n"); 5930 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5931 mActiveTrack->dump(buffer, SIZE); 5932 result.append(buffer); 5933 5934 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5935 result.append(buffer); 5936 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5937 result.append(buffer); 5938 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5939 result.append(buffer); 5940 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5941 result.append(buffer); 5942 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5943 result.append(buffer); 5944 5945 5946 } else { 5947 result.append("No record client\n"); 5948 } 5949 write(fd, result.string(), result.size()); 5950 5951 dumpBase(fd, args); 5952 dumpEffectChains(fd, args); 5953 5954 return NO_ERROR; 5955} 5956 5957// AudioBufferProvider interface 5958status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5959{ 5960 size_t framesReq = buffer->frameCount; 5961 size_t framesReady = mFrameCount - mRsmpInIndex; 5962 int channelCount; 5963 5964 if (framesReady == 0) { 5965 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5966 if (mBytesRead < 0) { 5967 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5968 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5969 // Force input into standby so that it tries to 5970 // recover at next read attempt 5971 mInput->stream->common.standby(&mInput->stream->common); 5972 usleep(kRecordThreadSleepUs); 5973 } 5974 buffer->raw = NULL; 5975 buffer->frameCount = 0; 5976 return NOT_ENOUGH_DATA; 5977 } 5978 mRsmpInIndex = 0; 5979 framesReady = mFrameCount; 5980 } 5981 5982 if (framesReq > framesReady) { 5983 framesReq = framesReady; 5984 } 5985 5986 if (mChannelCount == 1 && mReqChannelCount == 2) { 5987 channelCount = 1; 5988 } else { 5989 channelCount = 2; 5990 } 5991 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5992 buffer->frameCount = framesReq; 5993 return NO_ERROR; 5994} 5995 5996// AudioBufferProvider interface 5997void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5998{ 5999 mRsmpInIndex += buffer->frameCount; 6000 buffer->frameCount = 0; 6001} 6002 6003bool AudioFlinger::RecordThread::checkForNewParameters_l() 6004{ 6005 bool reconfig = false; 6006 6007 while (!mNewParameters.isEmpty()) { 6008 status_t status = NO_ERROR; 6009 String8 keyValuePair = mNewParameters[0]; 6010 AudioParameter param = AudioParameter(keyValuePair); 6011 int value; 6012 audio_format_t reqFormat = mFormat; 6013 int reqSamplingRate = mReqSampleRate; 6014 int reqChannelCount = mReqChannelCount; 6015 6016 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6017 reqSamplingRate = value; 6018 reconfig = true; 6019 } 6020 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6021 reqFormat = (audio_format_t) value; 6022 reconfig = true; 6023 } 6024 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6025 reqChannelCount = popcount(value); 6026 reconfig = true; 6027 } 6028 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6029 // do not accept frame count changes if tracks are open as the track buffer 6030 // size depends on frame count and correct behavior would not be guaranteed 6031 // if frame count is changed after track creation 6032 if (mActiveTrack != 0) { 6033 status = INVALID_OPERATION; 6034 } else { 6035 reconfig = true; 6036 } 6037 } 6038 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6039 // forward device change to effects that have requested to be 6040 // aware of attached audio device. 6041 for (size_t i = 0; i < mEffectChains.size(); i++) { 6042 mEffectChains[i]->setDevice_l(value); 6043 } 6044 // store input device and output device but do not forward output device to audio HAL. 6045 // Note that status is ignored by the caller for output device 6046 // (see AudioFlinger::setParameters() 6047 if (value & AUDIO_DEVICE_OUT_ALL) { 6048 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6049 status = BAD_VALUE; 6050 } else { 6051 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6052 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6053 if (mTrack != NULL) { 6054 bool suspend = audio_is_bluetooth_sco_device( 6055 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6056 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6057 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6058 } 6059 } 6060 mDevice |= (uint32_t)value; 6061 } 6062 if (status == NO_ERROR) { 6063 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6064 if (status == INVALID_OPERATION) { 6065 mInput->stream->common.standby(&mInput->stream->common); 6066 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6067 keyValuePair.string()); 6068 } 6069 if (reconfig) { 6070 if (status == BAD_VALUE && 6071 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6072 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6073 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6074 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6075 (reqChannelCount <= FCC_2)) { 6076 status = NO_ERROR; 6077 } 6078 if (status == NO_ERROR) { 6079 readInputParameters(); 6080 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6081 } 6082 } 6083 } 6084 6085 mNewParameters.removeAt(0); 6086 6087 mParamStatus = status; 6088 mParamCond.signal(); 6089 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6090 // already timed out waiting for the status and will never signal the condition. 6091 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6092 } 6093 return reconfig; 6094} 6095 6096String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6097{ 6098 char *s; 6099 String8 out_s8 = String8(); 6100 6101 Mutex::Autolock _l(mLock); 6102 if (initCheck() != NO_ERROR) { 6103 return out_s8; 6104 } 6105 6106 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6107 out_s8 = String8(s); 6108 free(s); 6109 return out_s8; 6110} 6111 6112void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6113 AudioSystem::OutputDescriptor desc; 6114 void *param2 = NULL; 6115 6116 switch (event) { 6117 case AudioSystem::INPUT_OPENED: 6118 case AudioSystem::INPUT_CONFIG_CHANGED: 6119 desc.channels = mChannelMask; 6120 desc.samplingRate = mSampleRate; 6121 desc.format = mFormat; 6122 desc.frameCount = mFrameCount; 6123 desc.latency = 0; 6124 param2 = &desc; 6125 break; 6126 6127 case AudioSystem::INPUT_CLOSED: 6128 default: 6129 break; 6130 } 6131 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6132} 6133 6134void AudioFlinger::RecordThread::readInputParameters() 6135{ 6136 delete mRsmpInBuffer; 6137 // mRsmpInBuffer is always assigned a new[] below 6138 delete mRsmpOutBuffer; 6139 mRsmpOutBuffer = NULL; 6140 delete mResampler; 6141 mResampler = NULL; 6142 6143 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6144 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6145 mChannelCount = (uint16_t)popcount(mChannelMask); 6146 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6147 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6148 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6149 mFrameCount = mInputBytes / mFrameSize; 6150 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6151 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6152 6153 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6154 { 6155 int channelCount; 6156 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6157 // stereo to mono post process as the resampler always outputs stereo. 6158 if (mChannelCount == 1 && mReqChannelCount == 2) { 6159 channelCount = 1; 6160 } else { 6161 channelCount = 2; 6162 } 6163 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6164 mResampler->setSampleRate(mSampleRate); 6165 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6166 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6167 6168 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6169 if (mChannelCount == 1 && mReqChannelCount == 1) { 6170 mFrameCount >>= 1; 6171 } 6172 6173 } 6174 mRsmpInIndex = mFrameCount; 6175} 6176 6177unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6178{ 6179 Mutex::Autolock _l(mLock); 6180 if (initCheck() != NO_ERROR) { 6181 return 0; 6182 } 6183 6184 return mInput->stream->get_input_frames_lost(mInput->stream); 6185} 6186 6187uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6188{ 6189 Mutex::Autolock _l(mLock); 6190 uint32_t result = 0; 6191 if (getEffectChain_l(sessionId) != 0) { 6192 result = EFFECT_SESSION; 6193 } 6194 6195 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6196 result |= TRACK_SESSION; 6197 } 6198 6199 return result; 6200} 6201 6202AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6203{ 6204 Mutex::Autolock _l(mLock); 6205 return mTrack; 6206} 6207 6208AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6209{ 6210 Mutex::Autolock _l(mLock); 6211 return mInput; 6212} 6213 6214AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6215{ 6216 Mutex::Autolock _l(mLock); 6217 AudioStreamIn *input = mInput; 6218 mInput = NULL; 6219 return input; 6220} 6221 6222// this method must always be called either with ThreadBase mLock held or inside the thread loop 6223audio_stream_t* AudioFlinger::RecordThread::stream() const 6224{ 6225 if (mInput == NULL) { 6226 return NULL; 6227 } 6228 return &mInput->stream->common; 6229} 6230 6231 6232// ---------------------------------------------------------------------------- 6233 6234audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6235{ 6236 if (!settingsAllowed()) { 6237 return 0; 6238 } 6239 Mutex::Autolock _l(mLock); 6240 return loadHwModule_l(name); 6241} 6242 6243// loadHwModule_l() must be called with AudioFlinger::mLock held 6244audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6245{ 6246 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6247 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6248 ALOGW("loadHwModule() module %s already loaded", name); 6249 return mAudioHwDevs.keyAt(i); 6250 } 6251 } 6252 6253 audio_hw_device_t *dev; 6254 6255 int rc = load_audio_interface(name, &dev); 6256 if (rc) { 6257 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6258 return 0; 6259 } 6260 6261 mHardwareStatus = AUDIO_HW_INIT; 6262 rc = dev->init_check(dev); 6263 mHardwareStatus = AUDIO_HW_IDLE; 6264 if (rc) { 6265 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6266 return 0; 6267 } 6268 6269 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6270 (NULL != dev->set_master_volume)) { 6271 AutoMutex lock(mHardwareLock); 6272 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6273 dev->set_master_volume(dev, mMasterVolume); 6274 mHardwareStatus = AUDIO_HW_IDLE; 6275 } 6276 6277 audio_module_handle_t handle = nextUniqueId(); 6278 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6279 6280 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6281 name, dev->common.module->name, dev->common.module->id, handle); 6282 6283 return handle; 6284 6285} 6286 6287audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6288 audio_devices_t *pDevices, 6289 uint32_t *pSamplingRate, 6290 audio_format_t *pFormat, 6291 audio_channel_mask_t *pChannelMask, 6292 uint32_t *pLatencyMs, 6293 audio_output_flags_t flags) 6294{ 6295 status_t status; 6296 PlaybackThread *thread = NULL; 6297 struct audio_config config = { 6298 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6299 channel_mask: pChannelMask ? *pChannelMask : 0, 6300 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6301 }; 6302 audio_stream_out_t *outStream = NULL; 6303 audio_hw_device_t *outHwDev; 6304 6305 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6306 module, 6307 (pDevices != NULL) ? (int)*pDevices : 0, 6308 config.sample_rate, 6309 config.format, 6310 config.channel_mask, 6311 flags); 6312 6313 if (pDevices == NULL || *pDevices == 0) { 6314 return 0; 6315 } 6316 6317 Mutex::Autolock _l(mLock); 6318 6319 outHwDev = findSuitableHwDev_l(module, *pDevices); 6320 if (outHwDev == NULL) 6321 return 0; 6322 6323 audio_io_handle_t id = nextUniqueId(); 6324 6325 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6326 6327 status = outHwDev->open_output_stream(outHwDev, 6328 id, 6329 *pDevices, 6330 (audio_output_flags_t)flags, 6331 &config, 6332 &outStream); 6333 6334 mHardwareStatus = AUDIO_HW_IDLE; 6335 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6336 outStream, 6337 config.sample_rate, 6338 config.format, 6339 config.channel_mask, 6340 status); 6341 6342 if (status == NO_ERROR && outStream != NULL) { 6343 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6344 6345 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6346 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6347 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6348 thread = new DirectOutputThread(this, output, id, *pDevices); 6349 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6350 } else { 6351 thread = new MixerThread(this, output, id, *pDevices); 6352 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6353 } 6354 mPlaybackThreads.add(id, thread); 6355 6356 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6357 if (pFormat != NULL) *pFormat = config.format; 6358 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6359 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6360 6361 // notify client processes of the new output creation 6362 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6363 6364 // the first primary output opened designates the primary hw device 6365 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6366 ALOGI("Using module %d has the primary audio interface", module); 6367 mPrimaryHardwareDev = outHwDev; 6368 6369 AutoMutex lock(mHardwareLock); 6370 mHardwareStatus = AUDIO_HW_SET_MODE; 6371 outHwDev->set_mode(outHwDev, mMode); 6372 6373 // Determine the level of master volume support the primary audio HAL has, 6374 // and set the initial master volume at the same time. 6375 float initialVolume = 1.0; 6376 mMasterVolumeSupportLvl = MVS_NONE; 6377 6378 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6379 if ((NULL != outHwDev->get_master_volume) && 6380 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6381 mMasterVolumeSupportLvl = MVS_FULL; 6382 } else { 6383 mMasterVolumeSupportLvl = MVS_SETONLY; 6384 initialVolume = 1.0; 6385 } 6386 6387 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6388 if ((NULL == outHwDev->set_master_volume) || 6389 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6390 mMasterVolumeSupportLvl = MVS_NONE; 6391 } 6392 // now that we have a primary device, initialize master volume on other devices 6393 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6394 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6395 6396 if ((dev != mPrimaryHardwareDev) && 6397 (NULL != dev->set_master_volume)) { 6398 dev->set_master_volume(dev, initialVolume); 6399 } 6400 } 6401 mHardwareStatus = AUDIO_HW_IDLE; 6402 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6403 ? initialVolume 6404 : 1.0; 6405 mMasterVolume = initialVolume; 6406 } 6407 return id; 6408 } 6409 6410 return 0; 6411} 6412 6413audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6414 audio_io_handle_t output2) 6415{ 6416 Mutex::Autolock _l(mLock); 6417 MixerThread *thread1 = checkMixerThread_l(output1); 6418 MixerThread *thread2 = checkMixerThread_l(output2); 6419 6420 if (thread1 == NULL || thread2 == NULL) { 6421 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6422 return 0; 6423 } 6424 6425 audio_io_handle_t id = nextUniqueId(); 6426 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6427 thread->addOutputTrack(thread2); 6428 mPlaybackThreads.add(id, thread); 6429 // notify client processes of the new output creation 6430 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6431 return id; 6432} 6433 6434status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6435{ 6436 // keep strong reference on the playback thread so that 6437 // it is not destroyed while exit() is executed 6438 sp<PlaybackThread> thread; 6439 { 6440 Mutex::Autolock _l(mLock); 6441 thread = checkPlaybackThread_l(output); 6442 if (thread == NULL) { 6443 return BAD_VALUE; 6444 } 6445 6446 ALOGV("closeOutput() %d", output); 6447 6448 if (thread->type() == ThreadBase::MIXER) { 6449 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6450 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6451 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6452 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6453 } 6454 } 6455 } 6456 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6457 mPlaybackThreads.removeItem(output); 6458 } 6459 thread->exit(); 6460 // The thread entity (active unit of execution) is no longer running here, 6461 // but the ThreadBase container still exists. 6462 6463 if (thread->type() != ThreadBase::DUPLICATING) { 6464 AudioStreamOut *out = thread->clearOutput(); 6465 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6466 // from now on thread->mOutput is NULL 6467 out->hwDev->close_output_stream(out->hwDev, out->stream); 6468 delete out; 6469 } 6470 return NO_ERROR; 6471} 6472 6473status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6474{ 6475 Mutex::Autolock _l(mLock); 6476 PlaybackThread *thread = checkPlaybackThread_l(output); 6477 6478 if (thread == NULL) { 6479 return BAD_VALUE; 6480 } 6481 6482 ALOGV("suspendOutput() %d", output); 6483 thread->suspend(); 6484 6485 return NO_ERROR; 6486} 6487 6488status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6489{ 6490 Mutex::Autolock _l(mLock); 6491 PlaybackThread *thread = checkPlaybackThread_l(output); 6492 6493 if (thread == NULL) { 6494 return BAD_VALUE; 6495 } 6496 6497 ALOGV("restoreOutput() %d", output); 6498 6499 thread->restore(); 6500 6501 return NO_ERROR; 6502} 6503 6504audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6505 audio_devices_t *pDevices, 6506 uint32_t *pSamplingRate, 6507 audio_format_t *pFormat, 6508 uint32_t *pChannelMask) 6509{ 6510 status_t status; 6511 RecordThread *thread = NULL; 6512 struct audio_config config = { 6513 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6514 channel_mask: pChannelMask ? *pChannelMask : 0, 6515 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6516 }; 6517 uint32_t reqSamplingRate = config.sample_rate; 6518 audio_format_t reqFormat = config.format; 6519 audio_channel_mask_t reqChannels = config.channel_mask; 6520 audio_stream_in_t *inStream = NULL; 6521 audio_hw_device_t *inHwDev; 6522 6523 if (pDevices == NULL || *pDevices == 0) { 6524 return 0; 6525 } 6526 6527 Mutex::Autolock _l(mLock); 6528 6529 inHwDev = findSuitableHwDev_l(module, *pDevices); 6530 if (inHwDev == NULL) 6531 return 0; 6532 6533 audio_io_handle_t id = nextUniqueId(); 6534 6535 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6536 &inStream); 6537 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6538 inStream, 6539 config.sample_rate, 6540 config.format, 6541 config.channel_mask, 6542 status); 6543 6544 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6545 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6546 // or stereo to mono conversions on 16 bit PCM inputs. 6547 if (status == BAD_VALUE && 6548 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6549 (config.sample_rate <= 2 * reqSamplingRate) && 6550 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6551 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6552 inStream = NULL; 6553 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6554 } 6555 6556 if (status == NO_ERROR && inStream != NULL) { 6557 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6558 6559 // Start record thread 6560 // RecorThread require both input and output device indication to forward to audio 6561 // pre processing modules 6562 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6563 thread = new RecordThread(this, 6564 input, 6565 reqSamplingRate, 6566 reqChannels, 6567 id, 6568 device); 6569 mRecordThreads.add(id, thread); 6570 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6571 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6572 if (pFormat != NULL) *pFormat = config.format; 6573 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6574 6575 input->stream->common.standby(&input->stream->common); 6576 6577 // notify client processes of the new input creation 6578 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6579 return id; 6580 } 6581 6582 return 0; 6583} 6584 6585status_t AudioFlinger::closeInput(audio_io_handle_t input) 6586{ 6587 // keep strong reference on the record thread so that 6588 // it is not destroyed while exit() is executed 6589 sp<RecordThread> thread; 6590 { 6591 Mutex::Autolock _l(mLock); 6592 thread = checkRecordThread_l(input); 6593 if (thread == NULL) { 6594 return BAD_VALUE; 6595 } 6596 6597 ALOGV("closeInput() %d", input); 6598 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6599 mRecordThreads.removeItem(input); 6600 } 6601 thread->exit(); 6602 // The thread entity (active unit of execution) is no longer running here, 6603 // but the ThreadBase container still exists. 6604 6605 AudioStreamIn *in = thread->clearInput(); 6606 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6607 // from now on thread->mInput is NULL 6608 in->hwDev->close_input_stream(in->hwDev, in->stream); 6609 delete in; 6610 6611 return NO_ERROR; 6612} 6613 6614status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6615{ 6616 Mutex::Autolock _l(mLock); 6617 MixerThread *dstThread = checkMixerThread_l(output); 6618 if (dstThread == NULL) { 6619 ALOGW("setStreamOutput() bad output id %d", output); 6620 return BAD_VALUE; 6621 } 6622 6623 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6624 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6625 6626 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6627 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6628 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6629 MixerThread *srcThread = (MixerThread *)thread; 6630 srcThread->invalidateTracks(stream); 6631 } 6632 } 6633 6634 return NO_ERROR; 6635} 6636 6637 6638int AudioFlinger::newAudioSessionId() 6639{ 6640 return nextUniqueId(); 6641} 6642 6643void AudioFlinger::acquireAudioSessionId(int audioSession) 6644{ 6645 Mutex::Autolock _l(mLock); 6646 pid_t caller = IPCThreadState::self()->getCallingPid(); 6647 ALOGV("acquiring %d from %d", audioSession, caller); 6648 size_t num = mAudioSessionRefs.size(); 6649 for (size_t i = 0; i< num; i++) { 6650 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6651 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6652 ref->mCnt++; 6653 ALOGV(" incremented refcount to %d", ref->mCnt); 6654 return; 6655 } 6656 } 6657 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6658 ALOGV(" added new entry for %d", audioSession); 6659} 6660 6661void AudioFlinger::releaseAudioSessionId(int audioSession) 6662{ 6663 Mutex::Autolock _l(mLock); 6664 pid_t caller = IPCThreadState::self()->getCallingPid(); 6665 ALOGV("releasing %d from %d", audioSession, caller); 6666 size_t num = mAudioSessionRefs.size(); 6667 for (size_t i = 0; i< num; i++) { 6668 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6669 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6670 ref->mCnt--; 6671 ALOGV(" decremented refcount to %d", ref->mCnt); 6672 if (ref->mCnt == 0) { 6673 mAudioSessionRefs.removeAt(i); 6674 delete ref; 6675 purgeStaleEffects_l(); 6676 } 6677 return; 6678 } 6679 } 6680 ALOGW("session id %d not found for pid %d", audioSession, caller); 6681} 6682 6683void AudioFlinger::purgeStaleEffects_l() { 6684 6685 ALOGV("purging stale effects"); 6686 6687 Vector< sp<EffectChain> > chains; 6688 6689 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6690 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6691 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6692 sp<EffectChain> ec = t->mEffectChains[j]; 6693 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6694 chains.push(ec); 6695 } 6696 } 6697 } 6698 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6699 sp<RecordThread> t = mRecordThreads.valueAt(i); 6700 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6701 sp<EffectChain> ec = t->mEffectChains[j]; 6702 chains.push(ec); 6703 } 6704 } 6705 6706 for (size_t i = 0; i < chains.size(); i++) { 6707 sp<EffectChain> ec = chains[i]; 6708 int sessionid = ec->sessionId(); 6709 sp<ThreadBase> t = ec->mThread.promote(); 6710 if (t == 0) { 6711 continue; 6712 } 6713 size_t numsessionrefs = mAudioSessionRefs.size(); 6714 bool found = false; 6715 for (size_t k = 0; k < numsessionrefs; k++) { 6716 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6717 if (ref->mSessionid == sessionid) { 6718 ALOGV(" session %d still exists for %d with %d refs", 6719 sessionid, ref->mPid, ref->mCnt); 6720 found = true; 6721 break; 6722 } 6723 } 6724 if (!found) { 6725 // remove all effects from the chain 6726 while (ec->mEffects.size()) { 6727 sp<EffectModule> effect = ec->mEffects[0]; 6728 effect->unPin(); 6729 Mutex::Autolock _l (t->mLock); 6730 t->removeEffect_l(effect); 6731 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6732 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6733 if (handle != 0) { 6734 handle->mEffect.clear(); 6735 if (handle->mHasControl && handle->mEnabled) { 6736 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6737 } 6738 } 6739 } 6740 AudioSystem::unregisterEffect(effect->id()); 6741 } 6742 } 6743 } 6744 return; 6745} 6746 6747// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6748AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6749{ 6750 return mPlaybackThreads.valueFor(output).get(); 6751} 6752 6753// checkMixerThread_l() must be called with AudioFlinger::mLock held 6754AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6755{ 6756 PlaybackThread *thread = checkPlaybackThread_l(output); 6757 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6758} 6759 6760// checkRecordThread_l() must be called with AudioFlinger::mLock held 6761AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6762{ 6763 return mRecordThreads.valueFor(input).get(); 6764} 6765 6766uint32_t AudioFlinger::nextUniqueId() 6767{ 6768 return android_atomic_inc(&mNextUniqueId); 6769} 6770 6771AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6772{ 6773 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6774 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6775 AudioStreamOut *output = thread->getOutput(); 6776 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6777 return thread; 6778 } 6779 } 6780 return NULL; 6781} 6782 6783uint32_t AudioFlinger::primaryOutputDevice_l() const 6784{ 6785 PlaybackThread *thread = primaryPlaybackThread_l(); 6786 6787 if (thread == NULL) { 6788 return 0; 6789 } 6790 6791 return thread->device(); 6792} 6793 6794sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6795 int triggerSession, 6796 int listenerSession, 6797 sync_event_callback_t callBack, 6798 void *cookie) 6799{ 6800 Mutex::Autolock _l(mLock); 6801 6802 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6803 status_t playStatus = NAME_NOT_FOUND; 6804 status_t recStatus = NAME_NOT_FOUND; 6805 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6806 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6807 if (playStatus == NO_ERROR) { 6808 return event; 6809 } 6810 } 6811 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6812 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6813 if (recStatus == NO_ERROR) { 6814 return event; 6815 } 6816 } 6817 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6818 mPendingSyncEvents.add(event); 6819 } else { 6820 ALOGV("createSyncEvent() invalid event %d", event->type()); 6821 event.clear(); 6822 } 6823 return event; 6824} 6825 6826// ---------------------------------------------------------------------------- 6827// Effect management 6828// ---------------------------------------------------------------------------- 6829 6830 6831status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6832{ 6833 Mutex::Autolock _l(mLock); 6834 return EffectQueryNumberEffects(numEffects); 6835} 6836 6837status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6838{ 6839 Mutex::Autolock _l(mLock); 6840 return EffectQueryEffect(index, descriptor); 6841} 6842 6843status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6844 effect_descriptor_t *descriptor) const 6845{ 6846 Mutex::Autolock _l(mLock); 6847 return EffectGetDescriptor(pUuid, descriptor); 6848} 6849 6850 6851sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6852 effect_descriptor_t *pDesc, 6853 const sp<IEffectClient>& effectClient, 6854 int32_t priority, 6855 audio_io_handle_t io, 6856 int sessionId, 6857 status_t *status, 6858 int *id, 6859 int *enabled) 6860{ 6861 status_t lStatus = NO_ERROR; 6862 sp<EffectHandle> handle; 6863 effect_descriptor_t desc; 6864 6865 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6866 pid, effectClient.get(), priority, sessionId, io); 6867 6868 if (pDesc == NULL) { 6869 lStatus = BAD_VALUE; 6870 goto Exit; 6871 } 6872 6873 // check audio settings permission for global effects 6874 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6875 lStatus = PERMISSION_DENIED; 6876 goto Exit; 6877 } 6878 6879 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6880 // that can only be created by audio policy manager (running in same process) 6881 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6882 lStatus = PERMISSION_DENIED; 6883 goto Exit; 6884 } 6885 6886 if (io == 0) { 6887 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6888 // output must be specified by AudioPolicyManager when using session 6889 // AUDIO_SESSION_OUTPUT_STAGE 6890 lStatus = BAD_VALUE; 6891 goto Exit; 6892 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6893 // if the output returned by getOutputForEffect() is removed before we lock the 6894 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6895 // and we will exit safely 6896 io = AudioSystem::getOutputForEffect(&desc); 6897 } 6898 } 6899 6900 { 6901 Mutex::Autolock _l(mLock); 6902 6903 6904 if (!EffectIsNullUuid(&pDesc->uuid)) { 6905 // if uuid is specified, request effect descriptor 6906 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6907 if (lStatus < 0) { 6908 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6909 goto Exit; 6910 } 6911 } else { 6912 // if uuid is not specified, look for an available implementation 6913 // of the required type in effect factory 6914 if (EffectIsNullUuid(&pDesc->type)) { 6915 ALOGW("createEffect() no effect type"); 6916 lStatus = BAD_VALUE; 6917 goto Exit; 6918 } 6919 uint32_t numEffects = 0; 6920 effect_descriptor_t d; 6921 d.flags = 0; // prevent compiler warning 6922 bool found = false; 6923 6924 lStatus = EffectQueryNumberEffects(&numEffects); 6925 if (lStatus < 0) { 6926 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6927 goto Exit; 6928 } 6929 for (uint32_t i = 0; i < numEffects; i++) { 6930 lStatus = EffectQueryEffect(i, &desc); 6931 if (lStatus < 0) { 6932 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6933 continue; 6934 } 6935 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6936 // If matching type found save effect descriptor. If the session is 6937 // 0 and the effect is not auxiliary, continue enumeration in case 6938 // an auxiliary version of this effect type is available 6939 found = true; 6940 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6941 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6942 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6943 break; 6944 } 6945 } 6946 } 6947 if (!found) { 6948 lStatus = BAD_VALUE; 6949 ALOGW("createEffect() effect not found"); 6950 goto Exit; 6951 } 6952 // For same effect type, chose auxiliary version over insert version if 6953 // connect to output mix (Compliance to OpenSL ES) 6954 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6955 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6956 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6957 } 6958 } 6959 6960 // Do not allow auxiliary effects on a session different from 0 (output mix) 6961 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6962 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6963 lStatus = INVALID_OPERATION; 6964 goto Exit; 6965 } 6966 6967 // check recording permission for visualizer 6968 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6969 !recordingAllowed()) { 6970 lStatus = PERMISSION_DENIED; 6971 goto Exit; 6972 } 6973 6974 // return effect descriptor 6975 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6976 6977 // If output is not specified try to find a matching audio session ID in one of the 6978 // output threads. 6979 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6980 // because of code checking output when entering the function. 6981 // Note: io is never 0 when creating an effect on an input 6982 if (io == 0) { 6983 // look for the thread where the specified audio session is present 6984 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6985 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6986 io = mPlaybackThreads.keyAt(i); 6987 break; 6988 } 6989 } 6990 if (io == 0) { 6991 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6992 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6993 io = mRecordThreads.keyAt(i); 6994 break; 6995 } 6996 } 6997 } 6998 // If no output thread contains the requested session ID, default to 6999 // first output. The effect chain will be moved to the correct output 7000 // thread when a track with the same session ID is created 7001 if (io == 0 && mPlaybackThreads.size()) { 7002 io = mPlaybackThreads.keyAt(0); 7003 } 7004 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7005 } 7006 ThreadBase *thread = checkRecordThread_l(io); 7007 if (thread == NULL) { 7008 thread = checkPlaybackThread_l(io); 7009 if (thread == NULL) { 7010 ALOGE("createEffect() unknown output thread"); 7011 lStatus = BAD_VALUE; 7012 goto Exit; 7013 } 7014 } 7015 7016 sp<Client> client = registerPid_l(pid); 7017 7018 // create effect on selected output thread 7019 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7020 &desc, enabled, &lStatus); 7021 if (handle != 0 && id != NULL) { 7022 *id = handle->id(); 7023 } 7024 } 7025 7026Exit: 7027 if (status != NULL) { 7028 *status = lStatus; 7029 } 7030 return handle; 7031} 7032 7033status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7034 audio_io_handle_t dstOutput) 7035{ 7036 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7037 sessionId, srcOutput, dstOutput); 7038 Mutex::Autolock _l(mLock); 7039 if (srcOutput == dstOutput) { 7040 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7041 return NO_ERROR; 7042 } 7043 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7044 if (srcThread == NULL) { 7045 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7046 return BAD_VALUE; 7047 } 7048 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7049 if (dstThread == NULL) { 7050 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7051 return BAD_VALUE; 7052 } 7053 7054 Mutex::Autolock _dl(dstThread->mLock); 7055 Mutex::Autolock _sl(srcThread->mLock); 7056 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7057 7058 return NO_ERROR; 7059} 7060 7061// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7062status_t AudioFlinger::moveEffectChain_l(int sessionId, 7063 AudioFlinger::PlaybackThread *srcThread, 7064 AudioFlinger::PlaybackThread *dstThread, 7065 bool reRegister) 7066{ 7067 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7068 sessionId, srcThread, dstThread); 7069 7070 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7071 if (chain == 0) { 7072 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7073 sessionId, srcThread); 7074 return INVALID_OPERATION; 7075 } 7076 7077 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7078 // so that a new chain is created with correct parameters when first effect is added. This is 7079 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7080 // removed. 7081 srcThread->removeEffectChain_l(chain); 7082 7083 // transfer all effects one by one so that new effect chain is created on new thread with 7084 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7085 audio_io_handle_t dstOutput = dstThread->id(); 7086 sp<EffectChain> dstChain; 7087 uint32_t strategy = 0; // prevent compiler warning 7088 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7089 while (effect != 0) { 7090 srcThread->removeEffect_l(effect); 7091 dstThread->addEffect_l(effect); 7092 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7093 if (effect->state() == EffectModule::ACTIVE || 7094 effect->state() == EffectModule::STOPPING) { 7095 effect->start(); 7096 } 7097 // if the move request is not received from audio policy manager, the effect must be 7098 // re-registered with the new strategy and output 7099 if (dstChain == 0) { 7100 dstChain = effect->chain().promote(); 7101 if (dstChain == 0) { 7102 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7103 srcThread->addEffect_l(effect); 7104 return NO_INIT; 7105 } 7106 strategy = dstChain->strategy(); 7107 } 7108 if (reRegister) { 7109 AudioSystem::unregisterEffect(effect->id()); 7110 AudioSystem::registerEffect(&effect->desc(), 7111 dstOutput, 7112 strategy, 7113 sessionId, 7114 effect->id()); 7115 } 7116 effect = chain->getEffectFromId_l(0); 7117 } 7118 7119 return NO_ERROR; 7120} 7121 7122 7123// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7124sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7125 const sp<AudioFlinger::Client>& client, 7126 const sp<IEffectClient>& effectClient, 7127 int32_t priority, 7128 int sessionId, 7129 effect_descriptor_t *desc, 7130 int *enabled, 7131 status_t *status 7132 ) 7133{ 7134 sp<EffectModule> effect; 7135 sp<EffectHandle> handle; 7136 status_t lStatus; 7137 sp<EffectChain> chain; 7138 bool chainCreated = false; 7139 bool effectCreated = false; 7140 bool effectRegistered = false; 7141 7142 lStatus = initCheck(); 7143 if (lStatus != NO_ERROR) { 7144 ALOGW("createEffect_l() Audio driver not initialized."); 7145 goto Exit; 7146 } 7147 7148 // Do not allow effects with session ID 0 on direct output or duplicating threads 7149 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7150 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7151 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7152 desc->name, sessionId); 7153 lStatus = BAD_VALUE; 7154 goto Exit; 7155 } 7156 // Only Pre processor effects are allowed on input threads and only on input threads 7157 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7158 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7159 desc->name, desc->flags, mType); 7160 lStatus = BAD_VALUE; 7161 goto Exit; 7162 } 7163 7164 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7165 7166 { // scope for mLock 7167 Mutex::Autolock _l(mLock); 7168 7169 // check for existing effect chain with the requested audio session 7170 chain = getEffectChain_l(sessionId); 7171 if (chain == 0) { 7172 // create a new chain for this session 7173 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7174 chain = new EffectChain(this, sessionId); 7175 addEffectChain_l(chain); 7176 chain->setStrategy(getStrategyForSession_l(sessionId)); 7177 chainCreated = true; 7178 } else { 7179 effect = chain->getEffectFromDesc_l(desc); 7180 } 7181 7182 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7183 7184 if (effect == 0) { 7185 int id = mAudioFlinger->nextUniqueId(); 7186 // Check CPU and memory usage 7187 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7188 if (lStatus != NO_ERROR) { 7189 goto Exit; 7190 } 7191 effectRegistered = true; 7192 // create a new effect module if none present in the chain 7193 effect = new EffectModule(this, chain, desc, id, sessionId); 7194 lStatus = effect->status(); 7195 if (lStatus != NO_ERROR) { 7196 goto Exit; 7197 } 7198 lStatus = chain->addEffect_l(effect); 7199 if (lStatus != NO_ERROR) { 7200 goto Exit; 7201 } 7202 effectCreated = true; 7203 7204 effect->setDevice(mDevice); 7205 effect->setMode(mAudioFlinger->getMode()); 7206 } 7207 // create effect handle and connect it to effect module 7208 handle = new EffectHandle(effect, client, effectClient, priority); 7209 lStatus = effect->addHandle(handle); 7210 if (enabled != NULL) { 7211 *enabled = (int)effect->isEnabled(); 7212 } 7213 } 7214 7215Exit: 7216 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7217 Mutex::Autolock _l(mLock); 7218 if (effectCreated) { 7219 chain->removeEffect_l(effect); 7220 } 7221 if (effectRegistered) { 7222 AudioSystem::unregisterEffect(effect->id()); 7223 } 7224 if (chainCreated) { 7225 removeEffectChain_l(chain); 7226 } 7227 handle.clear(); 7228 } 7229 7230 if (status != NULL) { 7231 *status = lStatus; 7232 } 7233 return handle; 7234} 7235 7236sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7237{ 7238 sp<EffectChain> chain = getEffectChain_l(sessionId); 7239 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7240} 7241 7242// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7243// PlaybackThread::mLock held 7244status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7245{ 7246 // check for existing effect chain with the requested audio session 7247 int sessionId = effect->sessionId(); 7248 sp<EffectChain> chain = getEffectChain_l(sessionId); 7249 bool chainCreated = false; 7250 7251 if (chain == 0) { 7252 // create a new chain for this session 7253 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7254 chain = new EffectChain(this, sessionId); 7255 addEffectChain_l(chain); 7256 chain->setStrategy(getStrategyForSession_l(sessionId)); 7257 chainCreated = true; 7258 } 7259 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7260 7261 if (chain->getEffectFromId_l(effect->id()) != 0) { 7262 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7263 this, effect->desc().name, chain.get()); 7264 return BAD_VALUE; 7265 } 7266 7267 status_t status = chain->addEffect_l(effect); 7268 if (status != NO_ERROR) { 7269 if (chainCreated) { 7270 removeEffectChain_l(chain); 7271 } 7272 return status; 7273 } 7274 7275 effect->setDevice(mDevice); 7276 effect->setMode(mAudioFlinger->getMode()); 7277 return NO_ERROR; 7278} 7279 7280void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7281 7282 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7283 effect_descriptor_t desc = effect->desc(); 7284 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7285 detachAuxEffect_l(effect->id()); 7286 } 7287 7288 sp<EffectChain> chain = effect->chain().promote(); 7289 if (chain != 0) { 7290 // remove effect chain if removing last effect 7291 if (chain->removeEffect_l(effect) == 0) { 7292 removeEffectChain_l(chain); 7293 } 7294 } else { 7295 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7296 } 7297} 7298 7299void AudioFlinger::ThreadBase::lockEffectChains_l( 7300 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7301{ 7302 effectChains = mEffectChains; 7303 for (size_t i = 0; i < mEffectChains.size(); i++) { 7304 mEffectChains[i]->lock(); 7305 } 7306} 7307 7308void AudioFlinger::ThreadBase::unlockEffectChains( 7309 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7310{ 7311 for (size_t i = 0; i < effectChains.size(); i++) { 7312 effectChains[i]->unlock(); 7313 } 7314} 7315 7316sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7317{ 7318 Mutex::Autolock _l(mLock); 7319 return getEffectChain_l(sessionId); 7320} 7321 7322sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7323{ 7324 size_t size = mEffectChains.size(); 7325 for (size_t i = 0; i < size; i++) { 7326 if (mEffectChains[i]->sessionId() == sessionId) { 7327 return mEffectChains[i]; 7328 } 7329 } 7330 return 0; 7331} 7332 7333void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7334{ 7335 Mutex::Autolock _l(mLock); 7336 size_t size = mEffectChains.size(); 7337 for (size_t i = 0; i < size; i++) { 7338 mEffectChains[i]->setMode_l(mode); 7339 } 7340} 7341 7342void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7343 const wp<EffectHandle>& handle, 7344 bool unpinIfLast) { 7345 7346 Mutex::Autolock _l(mLock); 7347 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7348 // delete the effect module if removing last handle on it 7349 if (effect->removeHandle(handle) == 0) { 7350 if (!effect->isPinned() || unpinIfLast) { 7351 removeEffect_l(effect); 7352 AudioSystem::unregisterEffect(effect->id()); 7353 } 7354 } 7355} 7356 7357status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7358{ 7359 int session = chain->sessionId(); 7360 int16_t *buffer = mMixBuffer; 7361 bool ownsBuffer = false; 7362 7363 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7364 if (session > 0) { 7365 // Only one effect chain can be present in direct output thread and it uses 7366 // the mix buffer as input 7367 if (mType != DIRECT) { 7368 size_t numSamples = mNormalFrameCount * mChannelCount; 7369 buffer = new int16_t[numSamples]; 7370 memset(buffer, 0, numSamples * sizeof(int16_t)); 7371 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7372 ownsBuffer = true; 7373 } 7374 7375 // Attach all tracks with same session ID to this chain. 7376 for (size_t i = 0; i < mTracks.size(); ++i) { 7377 sp<Track> track = mTracks[i]; 7378 if (session == track->sessionId()) { 7379 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7380 track->setMainBuffer(buffer); 7381 chain->incTrackCnt(); 7382 } 7383 } 7384 7385 // indicate all active tracks in the chain 7386 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7387 sp<Track> track = mActiveTracks[i].promote(); 7388 if (track == 0) continue; 7389 if (session == track->sessionId()) { 7390 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7391 chain->incActiveTrackCnt(); 7392 } 7393 } 7394 } 7395 7396 chain->setInBuffer(buffer, ownsBuffer); 7397 chain->setOutBuffer(mMixBuffer); 7398 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7399 // chains list in order to be processed last as it contains output stage effects 7400 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7401 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7402 // after track specific effects and before output stage 7403 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7404 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7405 // Effect chain for other sessions are inserted at beginning of effect 7406 // chains list to be processed before output mix effects. Relative order between other 7407 // sessions is not important 7408 size_t size = mEffectChains.size(); 7409 size_t i = 0; 7410 for (i = 0; i < size; i++) { 7411 if (mEffectChains[i]->sessionId() < session) break; 7412 } 7413 mEffectChains.insertAt(chain, i); 7414 checkSuspendOnAddEffectChain_l(chain); 7415 7416 return NO_ERROR; 7417} 7418 7419size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7420{ 7421 int session = chain->sessionId(); 7422 7423 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7424 7425 for (size_t i = 0; i < mEffectChains.size(); i++) { 7426 if (chain == mEffectChains[i]) { 7427 mEffectChains.removeAt(i); 7428 // detach all active tracks from the chain 7429 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7430 sp<Track> track = mActiveTracks[i].promote(); 7431 if (track == 0) continue; 7432 if (session == track->sessionId()) { 7433 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7434 chain.get(), session); 7435 chain->decActiveTrackCnt(); 7436 } 7437 } 7438 7439 // detach all tracks with same session ID from this chain 7440 for (size_t i = 0; i < mTracks.size(); ++i) { 7441 sp<Track> track = mTracks[i]; 7442 if (session == track->sessionId()) { 7443 track->setMainBuffer(mMixBuffer); 7444 chain->decTrackCnt(); 7445 } 7446 } 7447 break; 7448 } 7449 } 7450 return mEffectChains.size(); 7451} 7452 7453status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7454 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7455{ 7456 Mutex::Autolock _l(mLock); 7457 return attachAuxEffect_l(track, EffectId); 7458} 7459 7460status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7461 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7462{ 7463 status_t status = NO_ERROR; 7464 7465 if (EffectId == 0) { 7466 track->setAuxBuffer(0, NULL); 7467 } else { 7468 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7469 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7470 if (effect != 0) { 7471 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7472 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7473 } else { 7474 status = INVALID_OPERATION; 7475 } 7476 } else { 7477 status = BAD_VALUE; 7478 } 7479 } 7480 return status; 7481} 7482 7483void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7484{ 7485 for (size_t i = 0; i < mTracks.size(); ++i) { 7486 sp<Track> track = mTracks[i]; 7487 if (track->auxEffectId() == effectId) { 7488 attachAuxEffect_l(track, 0); 7489 } 7490 } 7491} 7492 7493status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7494{ 7495 // only one chain per input thread 7496 if (mEffectChains.size() != 0) { 7497 return INVALID_OPERATION; 7498 } 7499 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7500 7501 chain->setInBuffer(NULL); 7502 chain->setOutBuffer(NULL); 7503 7504 checkSuspendOnAddEffectChain_l(chain); 7505 7506 mEffectChains.add(chain); 7507 7508 return NO_ERROR; 7509} 7510 7511size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7512{ 7513 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7514 ALOGW_IF(mEffectChains.size() != 1, 7515 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7516 chain.get(), mEffectChains.size(), this); 7517 if (mEffectChains.size() == 1) { 7518 mEffectChains.removeAt(0); 7519 } 7520 return 0; 7521} 7522 7523// ---------------------------------------------------------------------------- 7524// EffectModule implementation 7525// ---------------------------------------------------------------------------- 7526 7527#undef LOG_TAG 7528#define LOG_TAG "AudioFlinger::EffectModule" 7529 7530AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7531 const wp<AudioFlinger::EffectChain>& chain, 7532 effect_descriptor_t *desc, 7533 int id, 7534 int sessionId) 7535 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7536 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7537{ 7538 ALOGV("Constructor %p", this); 7539 int lStatus; 7540 if (thread == NULL) { 7541 return; 7542 } 7543 7544 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7545 7546 // create effect engine from effect factory 7547 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7548 7549 if (mStatus != NO_ERROR) { 7550 return; 7551 } 7552 lStatus = init(); 7553 if (lStatus < 0) { 7554 mStatus = lStatus; 7555 goto Error; 7556 } 7557 7558 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7559 mPinned = true; 7560 } 7561 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7562 return; 7563Error: 7564 EffectRelease(mEffectInterface); 7565 mEffectInterface = NULL; 7566 ALOGV("Constructor Error %d", mStatus); 7567} 7568 7569AudioFlinger::EffectModule::~EffectModule() 7570{ 7571 ALOGV("Destructor %p", this); 7572 if (mEffectInterface != NULL) { 7573 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7574 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7575 sp<ThreadBase> thread = mThread.promote(); 7576 if (thread != 0) { 7577 audio_stream_t *stream = thread->stream(); 7578 if (stream != NULL) { 7579 stream->remove_audio_effect(stream, mEffectInterface); 7580 } 7581 } 7582 } 7583 // release effect engine 7584 EffectRelease(mEffectInterface); 7585 } 7586} 7587 7588status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7589{ 7590 status_t status; 7591 7592 Mutex::Autolock _l(mLock); 7593 int priority = handle->priority(); 7594 size_t size = mHandles.size(); 7595 sp<EffectHandle> h; 7596 size_t i; 7597 for (i = 0; i < size; i++) { 7598 h = mHandles[i].promote(); 7599 if (h == 0) continue; 7600 if (h->priority() <= priority) break; 7601 } 7602 // if inserted in first place, move effect control from previous owner to this handle 7603 if (i == 0) { 7604 bool enabled = false; 7605 if (h != 0) { 7606 enabled = h->enabled(); 7607 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7608 } 7609 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7610 status = NO_ERROR; 7611 } else { 7612 status = ALREADY_EXISTS; 7613 } 7614 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7615 mHandles.insertAt(handle, i); 7616 return status; 7617} 7618 7619size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7620{ 7621 Mutex::Autolock _l(mLock); 7622 size_t size = mHandles.size(); 7623 size_t i; 7624 for (i = 0; i < size; i++) { 7625 if (mHandles[i] == handle) break; 7626 } 7627 if (i == size) { 7628 return size; 7629 } 7630 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7631 7632 bool enabled = false; 7633 EffectHandle *hdl = handle.unsafe_get(); 7634 if (hdl != NULL) { 7635 ALOGV("removeHandle() unsafe_get OK"); 7636 enabled = hdl->enabled(); 7637 } 7638 mHandles.removeAt(i); 7639 size = mHandles.size(); 7640 // if removed from first place, move effect control from this handle to next in line 7641 if (i == 0 && size != 0) { 7642 sp<EffectHandle> h = mHandles[0].promote(); 7643 if (h != 0) { 7644 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7645 } 7646 } 7647 7648 // Prevent calls to process() and other functions on effect interface from now on. 7649 // The effect engine will be released by the destructor when the last strong reference on 7650 // this object is released which can happen after next process is called. 7651 if (size == 0 && !mPinned) { 7652 mState = DESTROYED; 7653 } 7654 7655 return size; 7656} 7657 7658sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7659{ 7660 Mutex::Autolock _l(mLock); 7661 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7662} 7663 7664void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7665{ 7666 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7667 // keep a strong reference on this EffectModule to avoid calling the 7668 // destructor before we exit 7669 sp<EffectModule> keep(this); 7670 { 7671 sp<ThreadBase> thread = mThread.promote(); 7672 if (thread != 0) { 7673 thread->disconnectEffect(keep, handle, unpinIfLast); 7674 } 7675 } 7676} 7677 7678void AudioFlinger::EffectModule::updateState() { 7679 Mutex::Autolock _l(mLock); 7680 7681 switch (mState) { 7682 case RESTART: 7683 reset_l(); 7684 // FALL THROUGH 7685 7686 case STARTING: 7687 // clear auxiliary effect input buffer for next accumulation 7688 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7689 memset(mConfig.inputCfg.buffer.raw, 7690 0, 7691 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7692 } 7693 start_l(); 7694 mState = ACTIVE; 7695 break; 7696 case STOPPING: 7697 stop_l(); 7698 mDisableWaitCnt = mMaxDisableWaitCnt; 7699 mState = STOPPED; 7700 break; 7701 case STOPPED: 7702 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7703 // turn off sequence. 7704 if (--mDisableWaitCnt == 0) { 7705 reset_l(); 7706 mState = IDLE; 7707 } 7708 break; 7709 default: //IDLE , ACTIVE, DESTROYED 7710 break; 7711 } 7712} 7713 7714void AudioFlinger::EffectModule::process() 7715{ 7716 Mutex::Autolock _l(mLock); 7717 7718 if (mState == DESTROYED || mEffectInterface == NULL || 7719 mConfig.inputCfg.buffer.raw == NULL || 7720 mConfig.outputCfg.buffer.raw == NULL) { 7721 return; 7722 } 7723 7724 if (isProcessEnabled()) { 7725 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7726 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7727 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7728 mConfig.inputCfg.buffer.s32, 7729 mConfig.inputCfg.buffer.frameCount/2); 7730 } 7731 7732 // do the actual processing in the effect engine 7733 int ret = (*mEffectInterface)->process(mEffectInterface, 7734 &mConfig.inputCfg.buffer, 7735 &mConfig.outputCfg.buffer); 7736 7737 // force transition to IDLE state when engine is ready 7738 if (mState == STOPPED && ret == -ENODATA) { 7739 mDisableWaitCnt = 1; 7740 } 7741 7742 // clear auxiliary effect input buffer for next accumulation 7743 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7744 memset(mConfig.inputCfg.buffer.raw, 0, 7745 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7746 } 7747 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7748 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7749 // If an insert effect is idle and input buffer is different from output buffer, 7750 // accumulate input onto output 7751 sp<EffectChain> chain = mChain.promote(); 7752 if (chain != 0 && chain->activeTrackCnt() != 0) { 7753 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7754 int16_t *in = mConfig.inputCfg.buffer.s16; 7755 int16_t *out = mConfig.outputCfg.buffer.s16; 7756 for (size_t i = 0; i < frameCnt; i++) { 7757 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7758 } 7759 } 7760 } 7761} 7762 7763void AudioFlinger::EffectModule::reset_l() 7764{ 7765 if (mEffectInterface == NULL) { 7766 return; 7767 } 7768 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7769} 7770 7771status_t AudioFlinger::EffectModule::configure() 7772{ 7773 uint32_t channels; 7774 if (mEffectInterface == NULL) { 7775 return NO_INIT; 7776 } 7777 7778 sp<ThreadBase> thread = mThread.promote(); 7779 if (thread == 0) { 7780 return DEAD_OBJECT; 7781 } 7782 7783 // TODO: handle configuration of effects replacing track process 7784 if (thread->channelCount() == 1) { 7785 channels = AUDIO_CHANNEL_OUT_MONO; 7786 } else { 7787 channels = AUDIO_CHANNEL_OUT_STEREO; 7788 } 7789 7790 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7791 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7792 } else { 7793 mConfig.inputCfg.channels = channels; 7794 } 7795 mConfig.outputCfg.channels = channels; 7796 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7797 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7798 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7799 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7800 mConfig.inputCfg.bufferProvider.cookie = NULL; 7801 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7802 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7803 mConfig.outputCfg.bufferProvider.cookie = NULL; 7804 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7805 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7806 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7807 // Insert effect: 7808 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7809 // always overwrites output buffer: input buffer == output buffer 7810 // - in other sessions: 7811 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7812 // other effect: overwrites output buffer: input buffer == output buffer 7813 // Auxiliary effect: 7814 // accumulates in output buffer: input buffer != output buffer 7815 // Therefore: accumulate <=> input buffer != output buffer 7816 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7817 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7818 } else { 7819 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7820 } 7821 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7822 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7823 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7824 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7825 7826 ALOGV("configure() %p thread %p buffer %p framecount %d", 7827 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7828 7829 status_t cmdStatus; 7830 uint32_t size = sizeof(int); 7831 status_t status = (*mEffectInterface)->command(mEffectInterface, 7832 EFFECT_CMD_SET_CONFIG, 7833 sizeof(effect_config_t), 7834 &mConfig, 7835 &size, 7836 &cmdStatus); 7837 if (status == 0) { 7838 status = cmdStatus; 7839 } 7840 7841 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7842 (1000 * mConfig.outputCfg.buffer.frameCount); 7843 7844 return status; 7845} 7846 7847status_t AudioFlinger::EffectModule::init() 7848{ 7849 Mutex::Autolock _l(mLock); 7850 if (mEffectInterface == NULL) { 7851 return NO_INIT; 7852 } 7853 status_t cmdStatus; 7854 uint32_t size = sizeof(status_t); 7855 status_t status = (*mEffectInterface)->command(mEffectInterface, 7856 EFFECT_CMD_INIT, 7857 0, 7858 NULL, 7859 &size, 7860 &cmdStatus); 7861 if (status == 0) { 7862 status = cmdStatus; 7863 } 7864 return status; 7865} 7866 7867status_t AudioFlinger::EffectModule::start() 7868{ 7869 Mutex::Autolock _l(mLock); 7870 return start_l(); 7871} 7872 7873status_t AudioFlinger::EffectModule::start_l() 7874{ 7875 if (mEffectInterface == NULL) { 7876 return NO_INIT; 7877 } 7878 status_t cmdStatus; 7879 uint32_t size = sizeof(status_t); 7880 status_t status = (*mEffectInterface)->command(mEffectInterface, 7881 EFFECT_CMD_ENABLE, 7882 0, 7883 NULL, 7884 &size, 7885 &cmdStatus); 7886 if (status == 0) { 7887 status = cmdStatus; 7888 } 7889 if (status == 0 && 7890 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7891 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7892 sp<ThreadBase> thread = mThread.promote(); 7893 if (thread != 0) { 7894 audio_stream_t *stream = thread->stream(); 7895 if (stream != NULL) { 7896 stream->add_audio_effect(stream, mEffectInterface); 7897 } 7898 } 7899 } 7900 return status; 7901} 7902 7903status_t AudioFlinger::EffectModule::stop() 7904{ 7905 Mutex::Autolock _l(mLock); 7906 return stop_l(); 7907} 7908 7909status_t AudioFlinger::EffectModule::stop_l() 7910{ 7911 if (mEffectInterface == NULL) { 7912 return NO_INIT; 7913 } 7914 status_t cmdStatus; 7915 uint32_t size = sizeof(status_t); 7916 status_t status = (*mEffectInterface)->command(mEffectInterface, 7917 EFFECT_CMD_DISABLE, 7918 0, 7919 NULL, 7920 &size, 7921 &cmdStatus); 7922 if (status == 0) { 7923 status = cmdStatus; 7924 } 7925 if (status == 0 && 7926 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7927 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7928 sp<ThreadBase> thread = mThread.promote(); 7929 if (thread != 0) { 7930 audio_stream_t *stream = thread->stream(); 7931 if (stream != NULL) { 7932 stream->remove_audio_effect(stream, mEffectInterface); 7933 } 7934 } 7935 } 7936 return status; 7937} 7938 7939status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7940 uint32_t cmdSize, 7941 void *pCmdData, 7942 uint32_t *replySize, 7943 void *pReplyData) 7944{ 7945 Mutex::Autolock _l(mLock); 7946// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7947 7948 if (mState == DESTROYED || mEffectInterface == NULL) { 7949 return NO_INIT; 7950 } 7951 status_t status = (*mEffectInterface)->command(mEffectInterface, 7952 cmdCode, 7953 cmdSize, 7954 pCmdData, 7955 replySize, 7956 pReplyData); 7957 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7958 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7959 for (size_t i = 1; i < mHandles.size(); i++) { 7960 sp<EffectHandle> h = mHandles[i].promote(); 7961 if (h != 0) { 7962 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7963 } 7964 } 7965 } 7966 return status; 7967} 7968 7969status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7970{ 7971 7972 Mutex::Autolock _l(mLock); 7973 ALOGV("setEnabled %p enabled %d", this, enabled); 7974 7975 if (enabled != isEnabled()) { 7976 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7977 if (enabled && status != NO_ERROR) { 7978 return status; 7979 } 7980 7981 switch (mState) { 7982 // going from disabled to enabled 7983 case IDLE: 7984 mState = STARTING; 7985 break; 7986 case STOPPED: 7987 mState = RESTART; 7988 break; 7989 case STOPPING: 7990 mState = ACTIVE; 7991 break; 7992 7993 // going from enabled to disabled 7994 case RESTART: 7995 mState = STOPPED; 7996 break; 7997 case STARTING: 7998 mState = IDLE; 7999 break; 8000 case ACTIVE: 8001 mState = STOPPING; 8002 break; 8003 case DESTROYED: 8004 return NO_ERROR; // simply ignore as we are being destroyed 8005 } 8006 for (size_t i = 1; i < mHandles.size(); i++) { 8007 sp<EffectHandle> h = mHandles[i].promote(); 8008 if (h != 0) { 8009 h->setEnabled(enabled); 8010 } 8011 } 8012 } 8013 return NO_ERROR; 8014} 8015 8016bool AudioFlinger::EffectModule::isEnabled() const 8017{ 8018 switch (mState) { 8019 case RESTART: 8020 case STARTING: 8021 case ACTIVE: 8022 return true; 8023 case IDLE: 8024 case STOPPING: 8025 case STOPPED: 8026 case DESTROYED: 8027 default: 8028 return false; 8029 } 8030} 8031 8032bool AudioFlinger::EffectModule::isProcessEnabled() const 8033{ 8034 switch (mState) { 8035 case RESTART: 8036 case ACTIVE: 8037 case STOPPING: 8038 case STOPPED: 8039 return true; 8040 case IDLE: 8041 case STARTING: 8042 case DESTROYED: 8043 default: 8044 return false; 8045 } 8046} 8047 8048status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8049{ 8050 Mutex::Autolock _l(mLock); 8051 status_t status = NO_ERROR; 8052 8053 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8054 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8055 if (isProcessEnabled() && 8056 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8057 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8058 status_t cmdStatus; 8059 uint32_t volume[2]; 8060 uint32_t *pVolume = NULL; 8061 uint32_t size = sizeof(volume); 8062 volume[0] = *left; 8063 volume[1] = *right; 8064 if (controller) { 8065 pVolume = volume; 8066 } 8067 status = (*mEffectInterface)->command(mEffectInterface, 8068 EFFECT_CMD_SET_VOLUME, 8069 size, 8070 volume, 8071 &size, 8072 pVolume); 8073 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8074 *left = volume[0]; 8075 *right = volume[1]; 8076 } 8077 } 8078 return status; 8079} 8080 8081status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8082{ 8083 Mutex::Autolock _l(mLock); 8084 status_t status = NO_ERROR; 8085 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8086 // audio pre processing modules on RecordThread can receive both output and 8087 // input device indication in the same call 8088 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8089 if (dev) { 8090 status_t cmdStatus; 8091 uint32_t size = sizeof(status_t); 8092 8093 status = (*mEffectInterface)->command(mEffectInterface, 8094 EFFECT_CMD_SET_DEVICE, 8095 sizeof(uint32_t), 8096 &dev, 8097 &size, 8098 &cmdStatus); 8099 if (status == NO_ERROR) { 8100 status = cmdStatus; 8101 } 8102 } 8103 dev = device & AUDIO_DEVICE_IN_ALL; 8104 if (dev) { 8105 status_t cmdStatus; 8106 uint32_t size = sizeof(status_t); 8107 8108 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8109 EFFECT_CMD_SET_INPUT_DEVICE, 8110 sizeof(uint32_t), 8111 &dev, 8112 &size, 8113 &cmdStatus); 8114 if (status2 == NO_ERROR) { 8115 status2 = cmdStatus; 8116 } 8117 if (status == NO_ERROR) { 8118 status = status2; 8119 } 8120 } 8121 } 8122 return status; 8123} 8124 8125status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8126{ 8127 Mutex::Autolock _l(mLock); 8128 status_t status = NO_ERROR; 8129 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8130 status_t cmdStatus; 8131 uint32_t size = sizeof(status_t); 8132 status = (*mEffectInterface)->command(mEffectInterface, 8133 EFFECT_CMD_SET_AUDIO_MODE, 8134 sizeof(audio_mode_t), 8135 &mode, 8136 &size, 8137 &cmdStatus); 8138 if (status == NO_ERROR) { 8139 status = cmdStatus; 8140 } 8141 } 8142 return status; 8143} 8144 8145void AudioFlinger::EffectModule::setSuspended(bool suspended) 8146{ 8147 Mutex::Autolock _l(mLock); 8148 mSuspended = suspended; 8149} 8150 8151bool AudioFlinger::EffectModule::suspended() const 8152{ 8153 Mutex::Autolock _l(mLock); 8154 return mSuspended; 8155} 8156 8157status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8158{ 8159 const size_t SIZE = 256; 8160 char buffer[SIZE]; 8161 String8 result; 8162 8163 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8164 result.append(buffer); 8165 8166 bool locked = tryLock(mLock); 8167 // failed to lock - AudioFlinger is probably deadlocked 8168 if (!locked) { 8169 result.append("\t\tCould not lock Fx mutex:\n"); 8170 } 8171 8172 result.append("\t\tSession Status State Engine:\n"); 8173 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8174 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8175 result.append(buffer); 8176 8177 result.append("\t\tDescriptor:\n"); 8178 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8179 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8180 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8181 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8182 result.append(buffer); 8183 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8184 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8185 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8186 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8187 result.append(buffer); 8188 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8189 mDescriptor.apiVersion, 8190 mDescriptor.flags); 8191 result.append(buffer); 8192 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8193 mDescriptor.name); 8194 result.append(buffer); 8195 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8196 mDescriptor.implementor); 8197 result.append(buffer); 8198 8199 result.append("\t\t- Input configuration:\n"); 8200 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8201 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8202 (uint32_t)mConfig.inputCfg.buffer.raw, 8203 mConfig.inputCfg.buffer.frameCount, 8204 mConfig.inputCfg.samplingRate, 8205 mConfig.inputCfg.channels, 8206 mConfig.inputCfg.format); 8207 result.append(buffer); 8208 8209 result.append("\t\t- Output configuration:\n"); 8210 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8211 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8212 (uint32_t)mConfig.outputCfg.buffer.raw, 8213 mConfig.outputCfg.buffer.frameCount, 8214 mConfig.outputCfg.samplingRate, 8215 mConfig.outputCfg.channels, 8216 mConfig.outputCfg.format); 8217 result.append(buffer); 8218 8219 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8220 result.append(buffer); 8221 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8222 for (size_t i = 0; i < mHandles.size(); ++i) { 8223 sp<EffectHandle> handle = mHandles[i].promote(); 8224 if (handle != 0) { 8225 handle->dump(buffer, SIZE); 8226 result.append(buffer); 8227 } 8228 } 8229 8230 result.append("\n"); 8231 8232 write(fd, result.string(), result.length()); 8233 8234 if (locked) { 8235 mLock.unlock(); 8236 } 8237 8238 return NO_ERROR; 8239} 8240 8241// ---------------------------------------------------------------------------- 8242// EffectHandle implementation 8243// ---------------------------------------------------------------------------- 8244 8245#undef LOG_TAG 8246#define LOG_TAG "AudioFlinger::EffectHandle" 8247 8248AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8249 const sp<AudioFlinger::Client>& client, 8250 const sp<IEffectClient>& effectClient, 8251 int32_t priority) 8252 : BnEffect(), 8253 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8254 mPriority(priority), mHasControl(false), mEnabled(false) 8255{ 8256 ALOGV("constructor %p", this); 8257 8258 if (client == 0) { 8259 return; 8260 } 8261 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8262 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8263 if (mCblkMemory != 0) { 8264 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8265 8266 if (mCblk != NULL) { 8267 new(mCblk) effect_param_cblk_t(); 8268 mBuffer = (uint8_t *)mCblk + bufOffset; 8269 } 8270 } else { 8271 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8272 return; 8273 } 8274} 8275 8276AudioFlinger::EffectHandle::~EffectHandle() 8277{ 8278 ALOGV("Destructor %p", this); 8279 disconnect(false); 8280 ALOGV("Destructor DONE %p", this); 8281} 8282 8283status_t AudioFlinger::EffectHandle::enable() 8284{ 8285 ALOGV("enable %p", this); 8286 if (!mHasControl) return INVALID_OPERATION; 8287 if (mEffect == 0) return DEAD_OBJECT; 8288 8289 if (mEnabled) { 8290 return NO_ERROR; 8291 } 8292 8293 mEnabled = true; 8294 8295 sp<ThreadBase> thread = mEffect->thread().promote(); 8296 if (thread != 0) { 8297 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8298 } 8299 8300 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8301 if (mEffect->suspended()) { 8302 return NO_ERROR; 8303 } 8304 8305 status_t status = mEffect->setEnabled(true); 8306 if (status != NO_ERROR) { 8307 if (thread != 0) { 8308 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8309 } 8310 mEnabled = false; 8311 } 8312 return status; 8313} 8314 8315status_t AudioFlinger::EffectHandle::disable() 8316{ 8317 ALOGV("disable %p", this); 8318 if (!mHasControl) return INVALID_OPERATION; 8319 if (mEffect == 0) return DEAD_OBJECT; 8320 8321 if (!mEnabled) { 8322 return NO_ERROR; 8323 } 8324 mEnabled = false; 8325 8326 if (mEffect->suspended()) { 8327 return NO_ERROR; 8328 } 8329 8330 status_t status = mEffect->setEnabled(false); 8331 8332 sp<ThreadBase> thread = mEffect->thread().promote(); 8333 if (thread != 0) { 8334 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8335 } 8336 8337 return status; 8338} 8339 8340void AudioFlinger::EffectHandle::disconnect() 8341{ 8342 disconnect(true); 8343} 8344 8345void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8346{ 8347 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8348 if (mEffect == 0) { 8349 return; 8350 } 8351 mEffect->disconnect(this, unpinIfLast); 8352 8353 if (mHasControl && mEnabled) { 8354 sp<ThreadBase> thread = mEffect->thread().promote(); 8355 if (thread != 0) { 8356 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8357 } 8358 } 8359 8360 // release sp on module => module destructor can be called now 8361 mEffect.clear(); 8362 if (mClient != 0) { 8363 if (mCblk != NULL) { 8364 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8365 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8366 } 8367 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8368 // Client destructor must run with AudioFlinger mutex locked 8369 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8370 mClient.clear(); 8371 } 8372} 8373 8374status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8375 uint32_t cmdSize, 8376 void *pCmdData, 8377 uint32_t *replySize, 8378 void *pReplyData) 8379{ 8380// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8381// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8382 8383 // only get parameter command is permitted for applications not controlling the effect 8384 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8385 return INVALID_OPERATION; 8386 } 8387 if (mEffect == 0) return DEAD_OBJECT; 8388 if (mClient == 0) return INVALID_OPERATION; 8389 8390 // handle commands that are not forwarded transparently to effect engine 8391 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8392 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8393 // no risk to block the whole media server process or mixer threads is we are stuck here 8394 Mutex::Autolock _l(mCblk->lock); 8395 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8396 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8397 mCblk->serverIndex = 0; 8398 mCblk->clientIndex = 0; 8399 return BAD_VALUE; 8400 } 8401 status_t status = NO_ERROR; 8402 while (mCblk->serverIndex < mCblk->clientIndex) { 8403 int reply; 8404 uint32_t rsize = sizeof(int); 8405 int *p = (int *)(mBuffer + mCblk->serverIndex); 8406 int size = *p++; 8407 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8408 ALOGW("command(): invalid parameter block size"); 8409 break; 8410 } 8411 effect_param_t *param = (effect_param_t *)p; 8412 if (param->psize == 0 || param->vsize == 0) { 8413 ALOGW("command(): null parameter or value size"); 8414 mCblk->serverIndex += size; 8415 continue; 8416 } 8417 uint32_t psize = sizeof(effect_param_t) + 8418 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8419 param->vsize; 8420 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8421 psize, 8422 p, 8423 &rsize, 8424 &reply); 8425 // stop at first error encountered 8426 if (ret != NO_ERROR) { 8427 status = ret; 8428 *(int *)pReplyData = reply; 8429 break; 8430 } else if (reply != NO_ERROR) { 8431 *(int *)pReplyData = reply; 8432 break; 8433 } 8434 mCblk->serverIndex += size; 8435 } 8436 mCblk->serverIndex = 0; 8437 mCblk->clientIndex = 0; 8438 return status; 8439 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8440 *(int *)pReplyData = NO_ERROR; 8441 return enable(); 8442 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8443 *(int *)pReplyData = NO_ERROR; 8444 return disable(); 8445 } 8446 8447 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8448} 8449 8450void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8451{ 8452 ALOGV("setControl %p control %d", this, hasControl); 8453 8454 mHasControl = hasControl; 8455 mEnabled = enabled; 8456 8457 if (signal && mEffectClient != 0) { 8458 mEffectClient->controlStatusChanged(hasControl); 8459 } 8460} 8461 8462void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8463 uint32_t cmdSize, 8464 void *pCmdData, 8465 uint32_t replySize, 8466 void *pReplyData) 8467{ 8468 if (mEffectClient != 0) { 8469 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8470 } 8471} 8472 8473 8474 8475void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8476{ 8477 if (mEffectClient != 0) { 8478 mEffectClient->enableStatusChanged(enabled); 8479 } 8480} 8481 8482status_t AudioFlinger::EffectHandle::onTransact( 8483 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8484{ 8485 return BnEffect::onTransact(code, data, reply, flags); 8486} 8487 8488 8489void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8490{ 8491 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8492 8493 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8494 (mClient == 0) ? getpid_cached : mClient->pid(), 8495 mPriority, 8496 mHasControl, 8497 !locked, 8498 mCblk ? mCblk->clientIndex : 0, 8499 mCblk ? mCblk->serverIndex : 0 8500 ); 8501 8502 if (locked) { 8503 mCblk->lock.unlock(); 8504 } 8505} 8506 8507#undef LOG_TAG 8508#define LOG_TAG "AudioFlinger::EffectChain" 8509 8510AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8511 int sessionId) 8512 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8513 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8514 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8515{ 8516 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8517 if (thread == NULL) { 8518 return; 8519 } 8520 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8521 thread->frameCount(); 8522} 8523 8524AudioFlinger::EffectChain::~EffectChain() 8525{ 8526 if (mOwnInBuffer) { 8527 delete mInBuffer; 8528 } 8529 8530} 8531 8532// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8533sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8534{ 8535 size_t size = mEffects.size(); 8536 8537 for (size_t i = 0; i < size; i++) { 8538 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8539 return mEffects[i]; 8540 } 8541 } 8542 return 0; 8543} 8544 8545// getEffectFromId_l() must be called with ThreadBase::mLock held 8546sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8547{ 8548 size_t size = mEffects.size(); 8549 8550 for (size_t i = 0; i < size; i++) { 8551 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8552 if (id == 0 || mEffects[i]->id() == id) { 8553 return mEffects[i]; 8554 } 8555 } 8556 return 0; 8557} 8558 8559// getEffectFromType_l() must be called with ThreadBase::mLock held 8560sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8561 const effect_uuid_t *type) 8562{ 8563 size_t size = mEffects.size(); 8564 8565 for (size_t i = 0; i < size; i++) { 8566 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8567 return mEffects[i]; 8568 } 8569 } 8570 return 0; 8571} 8572 8573// Must be called with EffectChain::mLock locked 8574void AudioFlinger::EffectChain::process_l() 8575{ 8576 sp<ThreadBase> thread = mThread.promote(); 8577 if (thread == 0) { 8578 ALOGW("process_l(): cannot promote mixer thread"); 8579 return; 8580 } 8581 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8582 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8583 // always process effects unless no more tracks are on the session and the effect tail 8584 // has been rendered 8585 bool doProcess = true; 8586 if (!isGlobalSession) { 8587 bool tracksOnSession = (trackCnt() != 0); 8588 8589 if (!tracksOnSession && mTailBufferCount == 0) { 8590 doProcess = false; 8591 } 8592 8593 if (activeTrackCnt() == 0) { 8594 // if no track is active and the effect tail has not been rendered, 8595 // the input buffer must be cleared here as the mixer process will not do it 8596 if (tracksOnSession || mTailBufferCount > 0) { 8597 size_t numSamples = thread->frameCount() * thread->channelCount(); 8598 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8599 if (mTailBufferCount > 0) { 8600 mTailBufferCount--; 8601 } 8602 } 8603 } 8604 } 8605 8606 size_t size = mEffects.size(); 8607 if (doProcess) { 8608 for (size_t i = 0; i < size; i++) { 8609 mEffects[i]->process(); 8610 } 8611 } 8612 for (size_t i = 0; i < size; i++) { 8613 mEffects[i]->updateState(); 8614 } 8615} 8616 8617// addEffect_l() must be called with PlaybackThread::mLock held 8618status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8619{ 8620 effect_descriptor_t desc = effect->desc(); 8621 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8622 8623 Mutex::Autolock _l(mLock); 8624 effect->setChain(this); 8625 sp<ThreadBase> thread = mThread.promote(); 8626 if (thread == 0) { 8627 return NO_INIT; 8628 } 8629 effect->setThread(thread); 8630 8631 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8632 // Auxiliary effects are inserted at the beginning of mEffects vector as 8633 // they are processed first and accumulated in chain input buffer 8634 mEffects.insertAt(effect, 0); 8635 8636 // the input buffer for auxiliary effect contains mono samples in 8637 // 32 bit format. This is to avoid saturation in AudoMixer 8638 // accumulation stage. Saturation is done in EffectModule::process() before 8639 // calling the process in effect engine 8640 size_t numSamples = thread->frameCount(); 8641 int32_t *buffer = new int32_t[numSamples]; 8642 memset(buffer, 0, numSamples * sizeof(int32_t)); 8643 effect->setInBuffer((int16_t *)buffer); 8644 // auxiliary effects output samples to chain input buffer for further processing 8645 // by insert effects 8646 effect->setOutBuffer(mInBuffer); 8647 } else { 8648 // Insert effects are inserted at the end of mEffects vector as they are processed 8649 // after track and auxiliary effects. 8650 // Insert effect order as a function of indicated preference: 8651 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8652 // another effect is present 8653 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8654 // last effect claiming first position 8655 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8656 // first effect claiming last position 8657 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8658 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8659 // already present 8660 8661 size_t size = mEffects.size(); 8662 size_t idx_insert = size; 8663 ssize_t idx_insert_first = -1; 8664 ssize_t idx_insert_last = -1; 8665 8666 for (size_t i = 0; i < size; i++) { 8667 effect_descriptor_t d = mEffects[i]->desc(); 8668 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8669 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8670 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8671 // check invalid effect chaining combinations 8672 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8673 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8674 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8675 return INVALID_OPERATION; 8676 } 8677 // remember position of first insert effect and by default 8678 // select this as insert position for new effect 8679 if (idx_insert == size) { 8680 idx_insert = i; 8681 } 8682 // remember position of last insert effect claiming 8683 // first position 8684 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8685 idx_insert_first = i; 8686 } 8687 // remember position of first insert effect claiming 8688 // last position 8689 if (iPref == EFFECT_FLAG_INSERT_LAST && 8690 idx_insert_last == -1) { 8691 idx_insert_last = i; 8692 } 8693 } 8694 } 8695 8696 // modify idx_insert from first position if needed 8697 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8698 if (idx_insert_last != -1) { 8699 idx_insert = idx_insert_last; 8700 } else { 8701 idx_insert = size; 8702 } 8703 } else { 8704 if (idx_insert_first != -1) { 8705 idx_insert = idx_insert_first + 1; 8706 } 8707 } 8708 8709 // always read samples from chain input buffer 8710 effect->setInBuffer(mInBuffer); 8711 8712 // if last effect in the chain, output samples to chain 8713 // output buffer, otherwise to chain input buffer 8714 if (idx_insert == size) { 8715 if (idx_insert != 0) { 8716 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8717 mEffects[idx_insert-1]->configure(); 8718 } 8719 effect->setOutBuffer(mOutBuffer); 8720 } else { 8721 effect->setOutBuffer(mInBuffer); 8722 } 8723 mEffects.insertAt(effect, idx_insert); 8724 8725 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8726 } 8727 effect->configure(); 8728 return NO_ERROR; 8729} 8730 8731// removeEffect_l() must be called with PlaybackThread::mLock held 8732size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8733{ 8734 Mutex::Autolock _l(mLock); 8735 size_t size = mEffects.size(); 8736 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8737 8738 for (size_t i = 0; i < size; i++) { 8739 if (effect == mEffects[i]) { 8740 // calling stop here will remove pre-processing effect from the audio HAL. 8741 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8742 // the middle of a read from audio HAL 8743 if (mEffects[i]->state() == EffectModule::ACTIVE || 8744 mEffects[i]->state() == EffectModule::STOPPING) { 8745 mEffects[i]->stop(); 8746 } 8747 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8748 delete[] effect->inBuffer(); 8749 } else { 8750 if (i == size - 1 && i != 0) { 8751 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8752 mEffects[i - 1]->configure(); 8753 } 8754 } 8755 mEffects.removeAt(i); 8756 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8757 break; 8758 } 8759 } 8760 8761 return mEffects.size(); 8762} 8763 8764// setDevice_l() must be called with PlaybackThread::mLock held 8765void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8766{ 8767 size_t size = mEffects.size(); 8768 for (size_t i = 0; i < size; i++) { 8769 mEffects[i]->setDevice(device); 8770 } 8771} 8772 8773// setMode_l() must be called with PlaybackThread::mLock held 8774void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8775{ 8776 size_t size = mEffects.size(); 8777 for (size_t i = 0; i < size; i++) { 8778 mEffects[i]->setMode(mode); 8779 } 8780} 8781 8782// setVolume_l() must be called with PlaybackThread::mLock held 8783bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8784{ 8785 uint32_t newLeft = *left; 8786 uint32_t newRight = *right; 8787 bool hasControl = false; 8788 int ctrlIdx = -1; 8789 size_t size = mEffects.size(); 8790 8791 // first update volume controller 8792 for (size_t i = size; i > 0; i--) { 8793 if (mEffects[i - 1]->isProcessEnabled() && 8794 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8795 ctrlIdx = i - 1; 8796 hasControl = true; 8797 break; 8798 } 8799 } 8800 8801 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8802 if (hasControl) { 8803 *left = mNewLeftVolume; 8804 *right = mNewRightVolume; 8805 } 8806 return hasControl; 8807 } 8808 8809 mVolumeCtrlIdx = ctrlIdx; 8810 mLeftVolume = newLeft; 8811 mRightVolume = newRight; 8812 8813 // second get volume update from volume controller 8814 if (ctrlIdx >= 0) { 8815 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8816 mNewLeftVolume = newLeft; 8817 mNewRightVolume = newRight; 8818 } 8819 // then indicate volume to all other effects in chain. 8820 // Pass altered volume to effects before volume controller 8821 // and requested volume to effects after controller 8822 uint32_t lVol = newLeft; 8823 uint32_t rVol = newRight; 8824 8825 for (size_t i = 0; i < size; i++) { 8826 if ((int)i == ctrlIdx) continue; 8827 // this also works for ctrlIdx == -1 when there is no volume controller 8828 if ((int)i > ctrlIdx) { 8829 lVol = *left; 8830 rVol = *right; 8831 } 8832 mEffects[i]->setVolume(&lVol, &rVol, false); 8833 } 8834 *left = newLeft; 8835 *right = newRight; 8836 8837 return hasControl; 8838} 8839 8840status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8841{ 8842 const size_t SIZE = 256; 8843 char buffer[SIZE]; 8844 String8 result; 8845 8846 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8847 result.append(buffer); 8848 8849 bool locked = tryLock(mLock); 8850 // failed to lock - AudioFlinger is probably deadlocked 8851 if (!locked) { 8852 result.append("\tCould not lock mutex:\n"); 8853 } 8854 8855 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8856 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8857 mEffects.size(), 8858 (uint32_t)mInBuffer, 8859 (uint32_t)mOutBuffer, 8860 mActiveTrackCnt); 8861 result.append(buffer); 8862 write(fd, result.string(), result.size()); 8863 8864 for (size_t i = 0; i < mEffects.size(); ++i) { 8865 sp<EffectModule> effect = mEffects[i]; 8866 if (effect != 0) { 8867 effect->dump(fd, args); 8868 } 8869 } 8870 8871 if (locked) { 8872 mLock.unlock(); 8873 } 8874 8875 return NO_ERROR; 8876} 8877 8878// must be called with ThreadBase::mLock held 8879void AudioFlinger::EffectChain::setEffectSuspended_l( 8880 const effect_uuid_t *type, bool suspend) 8881{ 8882 sp<SuspendedEffectDesc> desc; 8883 // use effect type UUID timelow as key as there is no real risk of identical 8884 // timeLow fields among effect type UUIDs. 8885 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8886 if (suspend) { 8887 if (index >= 0) { 8888 desc = mSuspendedEffects.valueAt(index); 8889 } else { 8890 desc = new SuspendedEffectDesc(); 8891 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8892 mSuspendedEffects.add(type->timeLow, desc); 8893 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8894 } 8895 if (desc->mRefCount++ == 0) { 8896 sp<EffectModule> effect = getEffectIfEnabled(type); 8897 if (effect != 0) { 8898 desc->mEffect = effect; 8899 effect->setSuspended(true); 8900 effect->setEnabled(false); 8901 } 8902 } 8903 } else { 8904 if (index < 0) { 8905 return; 8906 } 8907 desc = mSuspendedEffects.valueAt(index); 8908 if (desc->mRefCount <= 0) { 8909 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8910 desc->mRefCount = 1; 8911 } 8912 if (--desc->mRefCount == 0) { 8913 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8914 if (desc->mEffect != 0) { 8915 sp<EffectModule> effect = desc->mEffect.promote(); 8916 if (effect != 0) { 8917 effect->setSuspended(false); 8918 sp<EffectHandle> handle = effect->controlHandle(); 8919 if (handle != 0) { 8920 effect->setEnabled(handle->enabled()); 8921 } 8922 } 8923 desc->mEffect.clear(); 8924 } 8925 mSuspendedEffects.removeItemsAt(index); 8926 } 8927 } 8928} 8929 8930// must be called with ThreadBase::mLock held 8931void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8932{ 8933 sp<SuspendedEffectDesc> desc; 8934 8935 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8936 if (suspend) { 8937 if (index >= 0) { 8938 desc = mSuspendedEffects.valueAt(index); 8939 } else { 8940 desc = new SuspendedEffectDesc(); 8941 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8942 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8943 } 8944 if (desc->mRefCount++ == 0) { 8945 Vector< sp<EffectModule> > effects; 8946 getSuspendEligibleEffects(effects); 8947 for (size_t i = 0; i < effects.size(); i++) { 8948 setEffectSuspended_l(&effects[i]->desc().type, true); 8949 } 8950 } 8951 } else { 8952 if (index < 0) { 8953 return; 8954 } 8955 desc = mSuspendedEffects.valueAt(index); 8956 if (desc->mRefCount <= 0) { 8957 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8958 desc->mRefCount = 1; 8959 } 8960 if (--desc->mRefCount == 0) { 8961 Vector<const effect_uuid_t *> types; 8962 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8963 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8964 continue; 8965 } 8966 types.add(&mSuspendedEffects.valueAt(i)->mType); 8967 } 8968 for (size_t i = 0; i < types.size(); i++) { 8969 setEffectSuspended_l(types[i], false); 8970 } 8971 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8972 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8973 } 8974 } 8975} 8976 8977 8978// The volume effect is used for automated tests only 8979#ifndef OPENSL_ES_H_ 8980static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8981 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8982const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8983#endif //OPENSL_ES_H_ 8984 8985bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8986{ 8987 // auxiliary effects and visualizer are never suspended on output mix 8988 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8989 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8990 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8991 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8992 return false; 8993 } 8994 return true; 8995} 8996 8997void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8998{ 8999 effects.clear(); 9000 for (size_t i = 0; i < mEffects.size(); i++) { 9001 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9002 effects.add(mEffects[i]); 9003 } 9004 } 9005} 9006 9007sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9008 const effect_uuid_t *type) 9009{ 9010 sp<EffectModule> effect = getEffectFromType_l(type); 9011 return effect != 0 && effect->isEnabled() ? effect : 0; 9012} 9013 9014void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9015 bool enabled) 9016{ 9017 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9018 if (enabled) { 9019 if (index < 0) { 9020 // if the effect is not suspend check if all effects are suspended 9021 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9022 if (index < 0) { 9023 return; 9024 } 9025 if (!isEffectEligibleForSuspend(effect->desc())) { 9026 return; 9027 } 9028 setEffectSuspended_l(&effect->desc().type, enabled); 9029 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9030 if (index < 0) { 9031 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9032 return; 9033 } 9034 } 9035 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9036 effect->desc().type.timeLow); 9037 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9038 // if effect is requested to suspended but was not yet enabled, supend it now. 9039 if (desc->mEffect == 0) { 9040 desc->mEffect = effect; 9041 effect->setEnabled(false); 9042 effect->setSuspended(true); 9043 } 9044 } else { 9045 if (index < 0) { 9046 return; 9047 } 9048 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9049 effect->desc().type.timeLow); 9050 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9051 desc->mEffect.clear(); 9052 effect->setSuspended(false); 9053 } 9054} 9055 9056#undef LOG_TAG 9057#define LOG_TAG "AudioFlinger" 9058 9059// ---------------------------------------------------------------------------- 9060 9061status_t AudioFlinger::onTransact( 9062 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9063{ 9064 return BnAudioFlinger::onTransact(code, data, reply, flags); 9065} 9066 9067}; // namespace android 9068