AudioFlinger.cpp revision 300a2ee9327c05fbf9d3a5fd595b558097c7c5e8
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// Whether to use fast mixer 146static const enum { 147 FastMixer_Never, // never initialize or use: for debugging only 148 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 149 // normal mixer multiplier is 1 150 FastMixer_Static, // initialize if needed, then use all the time if initialized, 151 // multipler is calculated based on minimum normal mixer buffer size 152 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 153 // multipler is calculated based on minimum normal mixer buffer size 154 // FIXME for FastMixer_Dynamic: 155 // Supporting this option will require fixing HALs that can't handle large writes. 156 // For example, one HAL implementation returns an error from a large write, 157 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 158 // We could either fix the HAL implementations, or provide a wrapper that breaks 159 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 160} kUseFastMixer = FastMixer_Static; 161 162// ---------------------------------------------------------------------------- 163 164#ifdef ADD_BATTERY_DATA 165// To collect the amplifier usage 166static void addBatteryData(uint32_t params) { 167 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 168 if (service == NULL) { 169 // it already logged 170 return; 171 } 172 173 service->addBatteryData(params); 174} 175#endif 176 177static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 178{ 179 const hw_module_t *mod; 180 int rc; 181 182 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 183 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 184 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 185 if (rc) { 186 goto out; 187 } 188 rc = audio_hw_device_open(mod, dev); 189 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 190 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 191 if (rc) { 192 goto out; 193 } 194 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 195 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 196 rc = BAD_VALUE; 197 goto out; 198 } 199 return 0; 200 201out: 202 *dev = NULL; 203 return rc; 204} 205 206// ---------------------------------------------------------------------------- 207 208AudioFlinger::AudioFlinger() 209 : BnAudioFlinger(), 210 mPrimaryHardwareDev(NULL), 211 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 212 mMasterVolume(1.0f), 213 mMasterVolumeSupportLvl(MVS_NONE), 214 mMasterMute(false), 215 mNextUniqueId(1), 216 mMode(AUDIO_MODE_INVALID), 217 mBtNrecIsOff(false) 218{ 219} 220 221void AudioFlinger::onFirstRef() 222{ 223 int rc = 0; 224 225 Mutex::Autolock _l(mLock); 226 227 /* TODO: move all this work into an Init() function */ 228 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 229 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 230 uint32_t int_val; 231 if (1 == sscanf(val_str, "%u", &int_val)) { 232 mStandbyTimeInNsecs = milliseconds(int_val); 233 ALOGI("Using %u mSec as standby time.", int_val); 234 } else { 235 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 236 ALOGI("Using default %u mSec as standby time.", 237 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 238 } 239 } 240 241 mMode = AUDIO_MODE_NORMAL; 242 mMasterVolumeSW = 1.0; 243 mMasterVolume = 1.0; 244 mHardwareStatus = AUDIO_HW_IDLE; 245} 246 247AudioFlinger::~AudioFlinger() 248{ 249 250 while (!mRecordThreads.isEmpty()) { 251 // closeInput() will remove first entry from mRecordThreads 252 closeInput(mRecordThreads.keyAt(0)); 253 } 254 while (!mPlaybackThreads.isEmpty()) { 255 // closeOutput() will remove first entry from mPlaybackThreads 256 closeOutput(mPlaybackThreads.keyAt(0)); 257 } 258 259 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 260 // no mHardwareLock needed, as there are no other references to this 261 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 262 delete mAudioHwDevs.valueAt(i); 263 } 264} 265 266static const char * const audio_interfaces[] = { 267 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 268 AUDIO_HARDWARE_MODULE_ID_A2DP, 269 AUDIO_HARDWARE_MODULE_ID_USB, 270}; 271#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 272 273audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 274{ 275 // if module is 0, the request comes from an old policy manager and we should load 276 // well known modules 277 if (module == 0) { 278 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 279 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 280 loadHwModule_l(audio_interfaces[i]); 281 } 282 } else { 283 // check a match for the requested module handle 284 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 285 if (audioHwdevice != NULL) { 286 return audioHwdevice->hwDevice(); 287 } 288 } 289 // then try to find a module supporting the requested device. 290 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 291 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 292 if ((dev->get_supported_devices(dev) & devices) == devices) 293 return dev; 294 } 295 296 return NULL; 297} 298 299status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 300{ 301 const size_t SIZE = 256; 302 char buffer[SIZE]; 303 String8 result; 304 305 result.append("Clients:\n"); 306 for (size_t i = 0; i < mClients.size(); ++i) { 307 sp<Client> client = mClients.valueAt(i).promote(); 308 if (client != 0) { 309 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 310 result.append(buffer); 311 } 312 } 313 314 result.append("Global session refs:\n"); 315 result.append(" session pid count\n"); 316 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 317 AudioSessionRef *r = mAudioSessionRefs[i]; 318 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 319 result.append(buffer); 320 } 321 write(fd, result.string(), result.size()); 322 return NO_ERROR; 323} 324 325 326status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 327{ 328 const size_t SIZE = 256; 329 char buffer[SIZE]; 330 String8 result; 331 hardware_call_state hardwareStatus = mHardwareStatus; 332 333 snprintf(buffer, SIZE, "Hardware status: %d\n" 334 "Standby Time mSec: %u\n", 335 hardwareStatus, 336 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 337 result.append(buffer); 338 write(fd, result.string(), result.size()); 339 return NO_ERROR; 340} 341 342status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 343{ 344 const size_t SIZE = 256; 345 char buffer[SIZE]; 346 String8 result; 347 snprintf(buffer, SIZE, "Permission Denial: " 348 "can't dump AudioFlinger from pid=%d, uid=%d\n", 349 IPCThreadState::self()->getCallingPid(), 350 IPCThreadState::self()->getCallingUid()); 351 result.append(buffer); 352 write(fd, result.string(), result.size()); 353 return NO_ERROR; 354} 355 356static bool tryLock(Mutex& mutex) 357{ 358 bool locked = false; 359 for (int i = 0; i < kDumpLockRetries; ++i) { 360 if (mutex.tryLock() == NO_ERROR) { 361 locked = true; 362 break; 363 } 364 usleep(kDumpLockSleepUs); 365 } 366 return locked; 367} 368 369status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 370{ 371 if (!dumpAllowed()) { 372 dumpPermissionDenial(fd, args); 373 } else { 374 // get state of hardware lock 375 bool hardwareLocked = tryLock(mHardwareLock); 376 if (!hardwareLocked) { 377 String8 result(kHardwareLockedString); 378 write(fd, result.string(), result.size()); 379 } else { 380 mHardwareLock.unlock(); 381 } 382 383 bool locked = tryLock(mLock); 384 385 // failed to lock - AudioFlinger is probably deadlocked 386 if (!locked) { 387 String8 result(kDeadlockedString); 388 write(fd, result.string(), result.size()); 389 } 390 391 dumpClients(fd, args); 392 dumpInternals(fd, args); 393 394 // dump playback threads 395 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 396 mPlaybackThreads.valueAt(i)->dump(fd, args); 397 } 398 399 // dump record threads 400 for (size_t i = 0; i < mRecordThreads.size(); i++) { 401 mRecordThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump all hardware devs 405 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 406 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 407 dev->dump(dev, fd); 408 } 409 if (locked) mLock.unlock(); 410 } 411 return NO_ERROR; 412} 413 414sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 415{ 416 // If pid is already in the mClients wp<> map, then use that entry 417 // (for which promote() is always != 0), otherwise create a new entry and Client. 418 sp<Client> client = mClients.valueFor(pid).promote(); 419 if (client == 0) { 420 client = new Client(this, pid); 421 mClients.add(pid, client); 422 } 423 424 return client; 425} 426 427// IAudioFlinger interface 428 429 430sp<IAudioTrack> AudioFlinger::createTrack( 431 pid_t pid, 432 audio_stream_type_t streamType, 433 uint32_t sampleRate, 434 audio_format_t format, 435 uint32_t channelMask, 436 int frameCount, 437 IAudioFlinger::track_flags_t flags, 438 const sp<IMemory>& sharedBuffer, 439 audio_io_handle_t output, 440 pid_t tid, 441 int *sessionId, 442 status_t *status) 443{ 444 sp<PlaybackThread::Track> track; 445 sp<TrackHandle> trackHandle; 446 sp<Client> client; 447 status_t lStatus; 448 int lSessionId; 449 450 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 451 // but if someone uses binder directly they could bypass that and cause us to crash 452 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 453 ALOGE("createTrack() invalid stream type %d", streamType); 454 lStatus = BAD_VALUE; 455 goto Exit; 456 } 457 458 { 459 Mutex::Autolock _l(mLock); 460 PlaybackThread *thread = checkPlaybackThread_l(output); 461 PlaybackThread *effectThread = NULL; 462 if (thread == NULL) { 463 ALOGE("unknown output thread"); 464 lStatus = BAD_VALUE; 465 goto Exit; 466 } 467 468 client = registerPid_l(pid); 469 470 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 471 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 472 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 473 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 474 if (mPlaybackThreads.keyAt(i) != output) { 475 // prevent same audio session on different output threads 476 uint32_t sessions = t->hasAudioSession(*sessionId); 477 if (sessions & PlaybackThread::TRACK_SESSION) { 478 ALOGE("createTrack() session ID %d already in use", *sessionId); 479 lStatus = BAD_VALUE; 480 goto Exit; 481 } 482 // check if an effect with same session ID is waiting for a track to be created 483 if (sessions & PlaybackThread::EFFECT_SESSION) { 484 effectThread = t.get(); 485 } 486 } 487 } 488 lSessionId = *sessionId; 489 } else { 490 // if no audio session id is provided, create one here 491 lSessionId = nextUniqueId(); 492 if (sessionId != NULL) { 493 *sessionId = lSessionId; 494 } 495 } 496 ALOGV("createTrack() lSessionId: %d", lSessionId); 497 498 track = thread->createTrack_l(client, streamType, sampleRate, format, 499 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 500 501 // move effect chain to this output thread if an effect on same session was waiting 502 // for a track to be created 503 if (lStatus == NO_ERROR && effectThread != NULL) { 504 Mutex::Autolock _dl(thread->mLock); 505 Mutex::Autolock _sl(effectThread->mLock); 506 moveEffectChain_l(lSessionId, effectThread, thread, true); 507 } 508 509 // Look for sync events awaiting for a session to be used. 510 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 511 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 512 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 513 track->setSyncEvent(mPendingSyncEvents[i]); 514 mPendingSyncEvents.removeAt(i); 515 i--; 516 } 517 } 518 } 519 } 520 if (lStatus == NO_ERROR) { 521 trackHandle = new TrackHandle(track); 522 } else { 523 // remove local strong reference to Client before deleting the Track so that the Client 524 // destructor is called by the TrackBase destructor with mLock held 525 client.clear(); 526 track.clear(); 527 } 528 529Exit: 530 if (status != NULL) { 531 *status = lStatus; 532 } 533 return trackHandle; 534} 535 536uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 537{ 538 Mutex::Autolock _l(mLock); 539 PlaybackThread *thread = checkPlaybackThread_l(output); 540 if (thread == NULL) { 541 ALOGW("sampleRate() unknown thread %d", output); 542 return 0; 543 } 544 return thread->sampleRate(); 545} 546 547int AudioFlinger::channelCount(audio_io_handle_t output) const 548{ 549 Mutex::Autolock _l(mLock); 550 PlaybackThread *thread = checkPlaybackThread_l(output); 551 if (thread == NULL) { 552 ALOGW("channelCount() unknown thread %d", output); 553 return 0; 554 } 555 return thread->channelCount(); 556} 557 558audio_format_t AudioFlinger::format(audio_io_handle_t output) const 559{ 560 Mutex::Autolock _l(mLock); 561 PlaybackThread *thread = checkPlaybackThread_l(output); 562 if (thread == NULL) { 563 ALOGW("format() unknown thread %d", output); 564 return AUDIO_FORMAT_INVALID; 565 } 566 return thread->format(); 567} 568 569size_t AudioFlinger::frameCount(audio_io_handle_t output) const 570{ 571 Mutex::Autolock _l(mLock); 572 PlaybackThread *thread = checkPlaybackThread_l(output); 573 if (thread == NULL) { 574 ALOGW("frameCount() unknown thread %d", output); 575 return 0; 576 } 577 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 578 // should examine all callers and fix them to handle smaller counts 579 return thread->frameCount(); 580} 581 582uint32_t AudioFlinger::latency(audio_io_handle_t output) const 583{ 584 Mutex::Autolock _l(mLock); 585 PlaybackThread *thread = checkPlaybackThread_l(output); 586 if (thread == NULL) { 587 ALOGW("latency() unknown thread %d", output); 588 return 0; 589 } 590 return thread->latency(); 591} 592 593status_t AudioFlinger::setMasterVolume(float value) 594{ 595 status_t ret = initCheck(); 596 if (ret != NO_ERROR) { 597 return ret; 598 } 599 600 // check calling permissions 601 if (!settingsAllowed()) { 602 return PERMISSION_DENIED; 603 } 604 605 float swmv = value; 606 607 Mutex::Autolock _l(mLock); 608 609 // when hw supports master volume, don't scale in sw mixer 610 if (MVS_NONE != mMasterVolumeSupportLvl) { 611 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 612 AutoMutex lock(mHardwareLock); 613 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 614 615 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 616 if (NULL != dev->set_master_volume) { 617 dev->set_master_volume(dev, value); 618 } 619 mHardwareStatus = AUDIO_HW_IDLE; 620 } 621 622 swmv = 1.0; 623 } 624 625 mMasterVolume = value; 626 mMasterVolumeSW = swmv; 627 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 628 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 629 630 return NO_ERROR; 631} 632 633status_t AudioFlinger::setMode(audio_mode_t mode) 634{ 635 status_t ret = initCheck(); 636 if (ret != NO_ERROR) { 637 return ret; 638 } 639 640 // check calling permissions 641 if (!settingsAllowed()) { 642 return PERMISSION_DENIED; 643 } 644 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 645 ALOGW("Illegal value: setMode(%d)", mode); 646 return BAD_VALUE; 647 } 648 649 { // scope for the lock 650 AutoMutex lock(mHardwareLock); 651 mHardwareStatus = AUDIO_HW_SET_MODE; 652 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 653 mHardwareStatus = AUDIO_HW_IDLE; 654 } 655 656 if (NO_ERROR == ret) { 657 Mutex::Autolock _l(mLock); 658 mMode = mode; 659 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 660 mPlaybackThreads.valueAt(i)->setMode(mode); 661 } 662 663 return ret; 664} 665 666status_t AudioFlinger::setMicMute(bool state) 667{ 668 status_t ret = initCheck(); 669 if (ret != NO_ERROR) { 670 return ret; 671 } 672 673 // check calling permissions 674 if (!settingsAllowed()) { 675 return PERMISSION_DENIED; 676 } 677 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 680 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return ret; 683} 684 685bool AudioFlinger::getMicMute() const 686{ 687 status_t ret = initCheck(); 688 if (ret != NO_ERROR) { 689 return false; 690 } 691 692 bool state = AUDIO_MODE_INVALID; 693 AutoMutex lock(mHardwareLock); 694 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 695 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 696 mHardwareStatus = AUDIO_HW_IDLE; 697 return state; 698} 699 700status_t AudioFlinger::setMasterMute(bool muted) 701{ 702 // check calling permissions 703 if (!settingsAllowed()) { 704 return PERMISSION_DENIED; 705 } 706 707 Mutex::Autolock _l(mLock); 708 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 709 mMasterMute = muted; 710 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 711 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 712 713 return NO_ERROR; 714} 715 716float AudioFlinger::masterVolume() const 717{ 718 Mutex::Autolock _l(mLock); 719 return masterVolume_l(); 720} 721 722float AudioFlinger::masterVolumeSW() const 723{ 724 Mutex::Autolock _l(mLock); 725 return masterVolumeSW_l(); 726} 727 728bool AudioFlinger::masterMute() const 729{ 730 Mutex::Autolock _l(mLock); 731 return masterMute_l(); 732} 733 734float AudioFlinger::masterVolume_l() const 735{ 736 if (MVS_FULL == mMasterVolumeSupportLvl) { 737 float ret_val; 738 AutoMutex lock(mHardwareLock); 739 740 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 741 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 742 (NULL != mPrimaryHardwareDev->get_master_volume), 743 "can't get master volume"); 744 745 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 746 mHardwareStatus = AUDIO_HW_IDLE; 747 return ret_val; 748 } 749 750 return mMasterVolume; 751} 752 753status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 754 audio_io_handle_t output) 755{ 756 // check calling permissions 757 if (!settingsAllowed()) { 758 return PERMISSION_DENIED; 759 } 760 761 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 762 ALOGE("setStreamVolume() invalid stream %d", stream); 763 return BAD_VALUE; 764 } 765 766 AutoMutex lock(mLock); 767 PlaybackThread *thread = NULL; 768 if (output) { 769 thread = checkPlaybackThread_l(output); 770 if (thread == NULL) { 771 return BAD_VALUE; 772 } 773 } 774 775 mStreamTypes[stream].volume = value; 776 777 if (thread == NULL) { 778 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 779 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 780 } 781 } else { 782 thread->setStreamVolume(stream, value); 783 } 784 785 return NO_ERROR; 786} 787 788status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 789{ 790 // check calling permissions 791 if (!settingsAllowed()) { 792 return PERMISSION_DENIED; 793 } 794 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 796 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 797 ALOGE("setStreamMute() invalid stream %d", stream); 798 return BAD_VALUE; 799 } 800 801 AutoMutex lock(mLock); 802 mStreamTypes[stream].mute = muted; 803 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 804 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 805 806 return NO_ERROR; 807} 808 809float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 810{ 811 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 812 return 0.0f; 813 } 814 815 AutoMutex lock(mLock); 816 float volume; 817 if (output) { 818 PlaybackThread *thread = checkPlaybackThread_l(output); 819 if (thread == NULL) { 820 return 0.0f; 821 } 822 volume = thread->streamVolume(stream); 823 } else { 824 volume = streamVolume_l(stream); 825 } 826 827 return volume; 828} 829 830bool AudioFlinger::streamMute(audio_stream_type_t stream) const 831{ 832 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 833 return true; 834 } 835 836 AutoMutex lock(mLock); 837 return streamMute_l(stream); 838} 839 840status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 841{ 842 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 843 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 844 // check calling permissions 845 if (!settingsAllowed()) { 846 return PERMISSION_DENIED; 847 } 848 849 // ioHandle == 0 means the parameters are global to the audio hardware interface 850 if (ioHandle == 0) { 851 Mutex::Autolock _l(mLock); 852 status_t final_result = NO_ERROR; 853 { 854 AutoMutex lock(mHardwareLock); 855 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 856 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 857 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 858 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 859 final_result = result ?: final_result; 860 } 861 mHardwareStatus = AUDIO_HW_IDLE; 862 } 863 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 864 AudioParameter param = AudioParameter(keyValuePairs); 865 String8 value; 866 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 867 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 868 if (mBtNrecIsOff != btNrecIsOff) { 869 for (size_t i = 0; i < mRecordThreads.size(); i++) { 870 sp<RecordThread> thread = mRecordThreads.valueAt(i); 871 RecordThread::RecordTrack *track = thread->track(); 872 if (track != NULL) { 873 audio_devices_t device = (audio_devices_t)( 874 thread->device() & AUDIO_DEVICE_IN_ALL); 875 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 876 thread->setEffectSuspended(FX_IID_AEC, 877 suspend, 878 track->sessionId()); 879 thread->setEffectSuspended(FX_IID_NS, 880 suspend, 881 track->sessionId()); 882 } 883 } 884 mBtNrecIsOff = btNrecIsOff; 885 } 886 } 887 return final_result; 888 } 889 890 // hold a strong ref on thread in case closeOutput() or closeInput() is called 891 // and the thread is exited once the lock is released 892 sp<ThreadBase> thread; 893 { 894 Mutex::Autolock _l(mLock); 895 thread = checkPlaybackThread_l(ioHandle); 896 if (thread == NULL) { 897 thread = checkRecordThread_l(ioHandle); 898 } else if (thread == primaryPlaybackThread_l()) { 899 // indicate output device change to all input threads for pre processing 900 AudioParameter param = AudioParameter(keyValuePairs); 901 int value; 902 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 903 (value != 0)) { 904 for (size_t i = 0; i < mRecordThreads.size(); i++) { 905 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 906 } 907 } 908 } 909 } 910 if (thread != 0) { 911 return thread->setParameters(keyValuePairs); 912 } 913 return BAD_VALUE; 914} 915 916String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 917{ 918// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 919// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 920 921 Mutex::Autolock _l(mLock); 922 923 if (ioHandle == 0) { 924 String8 out_s8; 925 926 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 927 char *s; 928 { 929 AutoMutex lock(mHardwareLock); 930 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 931 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 932 s = dev->get_parameters(dev, keys.string()); 933 mHardwareStatus = AUDIO_HW_IDLE; 934 } 935 out_s8 += String8(s ? s : ""); 936 free(s); 937 } 938 return out_s8; 939 } 940 941 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 942 if (playbackThread != NULL) { 943 return playbackThread->getParameters(keys); 944 } 945 RecordThread *recordThread = checkRecordThread_l(ioHandle); 946 if (recordThread != NULL) { 947 return recordThread->getParameters(keys); 948 } 949 return String8(""); 950} 951 952size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 953{ 954 status_t ret = initCheck(); 955 if (ret != NO_ERROR) { 956 return 0; 957 } 958 959 AutoMutex lock(mHardwareLock); 960 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 961 struct audio_config config = { 962 sample_rate: sampleRate, 963 channel_mask: audio_channel_in_mask_from_count(channelCount), 964 format: format, 965 }; 966 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 967 mHardwareStatus = AUDIO_HW_IDLE; 968 return size; 969} 970 971unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 972{ 973 if (ioHandle == 0) { 974 return 0; 975 } 976 977 Mutex::Autolock _l(mLock); 978 979 RecordThread *recordThread = checkRecordThread_l(ioHandle); 980 if (recordThread != NULL) { 981 return recordThread->getInputFramesLost(); 982 } 983 return 0; 984} 985 986status_t AudioFlinger::setVoiceVolume(float value) 987{ 988 status_t ret = initCheck(); 989 if (ret != NO_ERROR) { 990 return ret; 991 } 992 993 // check calling permissions 994 if (!settingsAllowed()) { 995 return PERMISSION_DENIED; 996 } 997 998 AutoMutex lock(mHardwareLock); 999 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1000 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1001 mHardwareStatus = AUDIO_HW_IDLE; 1002 1003 return ret; 1004} 1005 1006status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1007 audio_io_handle_t output) const 1008{ 1009 status_t status; 1010 1011 Mutex::Autolock _l(mLock); 1012 1013 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1014 if (playbackThread != NULL) { 1015 return playbackThread->getRenderPosition(halFrames, dspFrames); 1016 } 1017 1018 return BAD_VALUE; 1019} 1020 1021void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1022{ 1023 1024 Mutex::Autolock _l(mLock); 1025 1026 pid_t pid = IPCThreadState::self()->getCallingPid(); 1027 if (mNotificationClients.indexOfKey(pid) < 0) { 1028 sp<NotificationClient> notificationClient = new NotificationClient(this, 1029 client, 1030 pid); 1031 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1032 1033 mNotificationClients.add(pid, notificationClient); 1034 1035 sp<IBinder> binder = client->asBinder(); 1036 binder->linkToDeath(notificationClient); 1037 1038 // the config change is always sent from playback or record threads to avoid deadlock 1039 // with AudioSystem::gLock 1040 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1041 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1042 } 1043 1044 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1045 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1046 } 1047 } 1048} 1049 1050void AudioFlinger::removeNotificationClient(pid_t pid) 1051{ 1052 Mutex::Autolock _l(mLock); 1053 1054 mNotificationClients.removeItem(pid); 1055 1056 ALOGV("%d died, releasing its sessions", pid); 1057 size_t num = mAudioSessionRefs.size(); 1058 bool removed = false; 1059 for (size_t i = 0; i< num; ) { 1060 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1061 ALOGV(" pid %d @ %d", ref->mPid, i); 1062 if (ref->mPid == pid) { 1063 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1064 mAudioSessionRefs.removeAt(i); 1065 delete ref; 1066 removed = true; 1067 num--; 1068 } else { 1069 i++; 1070 } 1071 } 1072 if (removed) { 1073 purgeStaleEffects_l(); 1074 } 1075} 1076 1077// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1078void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1079{ 1080 size_t size = mNotificationClients.size(); 1081 for (size_t i = 0; i < size; i++) { 1082 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1083 param2); 1084 } 1085} 1086 1087// removeClient_l() must be called with AudioFlinger::mLock held 1088void AudioFlinger::removeClient_l(pid_t pid) 1089{ 1090 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1091 mClients.removeItem(pid); 1092} 1093 1094 1095// ---------------------------------------------------------------------------- 1096 1097AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1098 uint32_t device, type_t type) 1099 : Thread(false), 1100 mType(type), 1101 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1102 // mChannelMask 1103 mChannelCount(0), 1104 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1105 mParamStatus(NO_ERROR), 1106 mStandby(false), mId(id), 1107 mDevice(device), 1108 mDeathRecipient(new PMDeathRecipient(this)) 1109{ 1110} 1111 1112AudioFlinger::ThreadBase::~ThreadBase() 1113{ 1114 mParamCond.broadcast(); 1115 // do not lock the mutex in destructor 1116 releaseWakeLock_l(); 1117 if (mPowerManager != 0) { 1118 sp<IBinder> binder = mPowerManager->asBinder(); 1119 binder->unlinkToDeath(mDeathRecipient); 1120 } 1121} 1122 1123void AudioFlinger::ThreadBase::exit() 1124{ 1125 ALOGV("ThreadBase::exit"); 1126 { 1127 // This lock prevents the following race in thread (uniprocessor for illustration): 1128 // if (!exitPending()) { 1129 // // context switch from here to exit() 1130 // // exit() calls requestExit(), what exitPending() observes 1131 // // exit() calls signal(), which is dropped since no waiters 1132 // // context switch back from exit() to here 1133 // mWaitWorkCV.wait(...); 1134 // // now thread is hung 1135 // } 1136 AutoMutex lock(mLock); 1137 requestExit(); 1138 mWaitWorkCV.signal(); 1139 } 1140 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1141 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1142 requestExitAndWait(); 1143} 1144 1145status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1146{ 1147 status_t status; 1148 1149 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1150 Mutex::Autolock _l(mLock); 1151 1152 mNewParameters.add(keyValuePairs); 1153 mWaitWorkCV.signal(); 1154 // wait condition with timeout in case the thread loop has exited 1155 // before the request could be processed 1156 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1157 status = mParamStatus; 1158 mWaitWorkCV.signal(); 1159 } else { 1160 status = TIMED_OUT; 1161 } 1162 return status; 1163} 1164 1165void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1166{ 1167 Mutex::Autolock _l(mLock); 1168 sendConfigEvent_l(event, param); 1169} 1170 1171// sendConfigEvent_l() must be called with ThreadBase::mLock held 1172void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1173{ 1174 ConfigEvent configEvent; 1175 configEvent.mEvent = event; 1176 configEvent.mParam = param; 1177 mConfigEvents.add(configEvent); 1178 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1179 mWaitWorkCV.signal(); 1180} 1181 1182void AudioFlinger::ThreadBase::processConfigEvents() 1183{ 1184 mLock.lock(); 1185 while (!mConfigEvents.isEmpty()) { 1186 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1187 ConfigEvent configEvent = mConfigEvents[0]; 1188 mConfigEvents.removeAt(0); 1189 // release mLock before locking AudioFlinger mLock: lock order is always 1190 // AudioFlinger then ThreadBase to avoid cross deadlock 1191 mLock.unlock(); 1192 mAudioFlinger->mLock.lock(); 1193 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1194 mAudioFlinger->mLock.unlock(); 1195 mLock.lock(); 1196 } 1197 mLock.unlock(); 1198} 1199 1200status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1201{ 1202 const size_t SIZE = 256; 1203 char buffer[SIZE]; 1204 String8 result; 1205 1206 bool locked = tryLock(mLock); 1207 if (!locked) { 1208 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1209 write(fd, buffer, strlen(buffer)); 1210 } 1211 1212 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1213 result.append(buffer); 1214 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1217 result.append(buffer); 1218 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1219 result.append(buffer); 1220 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1221 result.append(buffer); 1222 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1223 result.append(buffer); 1224 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1225 result.append(buffer); 1226 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1227 result.append(buffer); 1228 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1229 result.append(buffer); 1230 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1231 result.append(buffer); 1232 1233 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1234 result.append(buffer); 1235 result.append(" Index Command"); 1236 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1237 snprintf(buffer, SIZE, "\n %02d ", i); 1238 result.append(buffer); 1239 result.append(mNewParameters[i]); 1240 } 1241 1242 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1243 result.append(buffer); 1244 snprintf(buffer, SIZE, " Index event param\n"); 1245 result.append(buffer); 1246 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1247 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1248 result.append(buffer); 1249 } 1250 result.append("\n"); 1251 1252 write(fd, result.string(), result.size()); 1253 1254 if (locked) { 1255 mLock.unlock(); 1256 } 1257 return NO_ERROR; 1258} 1259 1260status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1261{ 1262 const size_t SIZE = 256; 1263 char buffer[SIZE]; 1264 String8 result; 1265 1266 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1267 write(fd, buffer, strlen(buffer)); 1268 1269 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1270 sp<EffectChain> chain = mEffectChains[i]; 1271 if (chain != 0) { 1272 chain->dump(fd, args); 1273 } 1274 } 1275 return NO_ERROR; 1276} 1277 1278void AudioFlinger::ThreadBase::acquireWakeLock() 1279{ 1280 Mutex::Autolock _l(mLock); 1281 acquireWakeLock_l(); 1282} 1283 1284void AudioFlinger::ThreadBase::acquireWakeLock_l() 1285{ 1286 if (mPowerManager == 0) { 1287 // use checkService() to avoid blocking if power service is not up yet 1288 sp<IBinder> binder = 1289 defaultServiceManager()->checkService(String16("power")); 1290 if (binder == 0) { 1291 ALOGW("Thread %s cannot connect to the power manager service", mName); 1292 } else { 1293 mPowerManager = interface_cast<IPowerManager>(binder); 1294 binder->linkToDeath(mDeathRecipient); 1295 } 1296 } 1297 if (mPowerManager != 0) { 1298 sp<IBinder> binder = new BBinder(); 1299 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1300 binder, 1301 String16(mName)); 1302 if (status == NO_ERROR) { 1303 mWakeLockToken = binder; 1304 } 1305 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1306 } 1307} 1308 1309void AudioFlinger::ThreadBase::releaseWakeLock() 1310{ 1311 Mutex::Autolock _l(mLock); 1312 releaseWakeLock_l(); 1313} 1314 1315void AudioFlinger::ThreadBase::releaseWakeLock_l() 1316{ 1317 if (mWakeLockToken != 0) { 1318 ALOGV("releaseWakeLock_l() %s", mName); 1319 if (mPowerManager != 0) { 1320 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1321 } 1322 mWakeLockToken.clear(); 1323 } 1324} 1325 1326void AudioFlinger::ThreadBase::clearPowerManager() 1327{ 1328 Mutex::Autolock _l(mLock); 1329 releaseWakeLock_l(); 1330 mPowerManager.clear(); 1331} 1332 1333void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1334{ 1335 sp<ThreadBase> thread = mThread.promote(); 1336 if (thread != 0) { 1337 thread->clearPowerManager(); 1338 } 1339 ALOGW("power manager service died !!!"); 1340} 1341 1342void AudioFlinger::ThreadBase::setEffectSuspended( 1343 const effect_uuid_t *type, bool suspend, int sessionId) 1344{ 1345 Mutex::Autolock _l(mLock); 1346 setEffectSuspended_l(type, suspend, sessionId); 1347} 1348 1349void AudioFlinger::ThreadBase::setEffectSuspended_l( 1350 const effect_uuid_t *type, bool suspend, int sessionId) 1351{ 1352 sp<EffectChain> chain = getEffectChain_l(sessionId); 1353 if (chain != 0) { 1354 if (type != NULL) { 1355 chain->setEffectSuspended_l(type, suspend); 1356 } else { 1357 chain->setEffectSuspendedAll_l(suspend); 1358 } 1359 } 1360 1361 updateSuspendedSessions_l(type, suspend, sessionId); 1362} 1363 1364void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1365{ 1366 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1367 if (index < 0) { 1368 return; 1369 } 1370 1371 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1372 mSuspendedSessions.editValueAt(index); 1373 1374 for (size_t i = 0; i < sessionEffects.size(); i++) { 1375 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1376 for (int j = 0; j < desc->mRefCount; j++) { 1377 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1378 chain->setEffectSuspendedAll_l(true); 1379 } else { 1380 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1381 desc->mType.timeLow); 1382 chain->setEffectSuspended_l(&desc->mType, true); 1383 } 1384 } 1385 } 1386} 1387 1388void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1389 bool suspend, 1390 int sessionId) 1391{ 1392 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1393 1394 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1395 1396 if (suspend) { 1397 if (index >= 0) { 1398 sessionEffects = mSuspendedSessions.editValueAt(index); 1399 } else { 1400 mSuspendedSessions.add(sessionId, sessionEffects); 1401 } 1402 } else { 1403 if (index < 0) { 1404 return; 1405 } 1406 sessionEffects = mSuspendedSessions.editValueAt(index); 1407 } 1408 1409 1410 int key = EffectChain::kKeyForSuspendAll; 1411 if (type != NULL) { 1412 key = type->timeLow; 1413 } 1414 index = sessionEffects.indexOfKey(key); 1415 1416 sp<SuspendedSessionDesc> desc; 1417 if (suspend) { 1418 if (index >= 0) { 1419 desc = sessionEffects.valueAt(index); 1420 } else { 1421 desc = new SuspendedSessionDesc(); 1422 if (type != NULL) { 1423 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1424 } 1425 sessionEffects.add(key, desc); 1426 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1427 } 1428 desc->mRefCount++; 1429 } else { 1430 if (index < 0) { 1431 return; 1432 } 1433 desc = sessionEffects.valueAt(index); 1434 if (--desc->mRefCount == 0) { 1435 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1436 sessionEffects.removeItemsAt(index); 1437 if (sessionEffects.isEmpty()) { 1438 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1439 sessionId); 1440 mSuspendedSessions.removeItem(sessionId); 1441 } 1442 } 1443 } 1444 if (!sessionEffects.isEmpty()) { 1445 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1446 } 1447} 1448 1449void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1450 bool enabled, 1451 int sessionId) 1452{ 1453 Mutex::Autolock _l(mLock); 1454 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1455} 1456 1457void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1458 bool enabled, 1459 int sessionId) 1460{ 1461 if (mType != RECORD) { 1462 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1463 // another session. This gives the priority to well behaved effect control panels 1464 // and applications not using global effects. 1465 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1466 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1467 } 1468 } 1469 1470 sp<EffectChain> chain = getEffectChain_l(sessionId); 1471 if (chain != 0) { 1472 chain->checkSuspendOnEffectEnabled(effect, enabled); 1473 } 1474} 1475 1476// ---------------------------------------------------------------------------- 1477 1478AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1479 AudioStreamOut* output, 1480 audio_io_handle_t id, 1481 uint32_t device, 1482 type_t type) 1483 : ThreadBase(audioFlinger, id, device, type), 1484 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1485 // Assumes constructor is called by AudioFlinger with it's mLock held, 1486 // but it would be safer to explicitly pass initial masterMute as parameter 1487 mMasterMute(audioFlinger->masterMute_l()), 1488 // mStreamTypes[] initialized in constructor body 1489 mOutput(output), 1490 // Assumes constructor is called by AudioFlinger with it's mLock held, 1491 // but it would be safer to explicitly pass initial masterVolume as parameter 1492 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1493 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1494 mMixerStatus(MIXER_IDLE), 1495 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1496 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1497 mFastTrackNewMask(0) 1498{ 1499#if !LOG_NDEBUG 1500 memset(mFastTrackNewArray, 0, sizeof(mFastTrackNewArray)); 1501#endif 1502 snprintf(mName, kNameLength, "AudioOut_%X", id); 1503 1504 readOutputParameters(); 1505 1506 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1507 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1508 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1509 stream = (audio_stream_type_t) (stream + 1)) { 1510 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1511 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1512 } 1513 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1514 // because mAudioFlinger doesn't have one to copy from 1515} 1516 1517AudioFlinger::PlaybackThread::~PlaybackThread() 1518{ 1519 delete [] mMixBuffer; 1520} 1521 1522status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1523{ 1524 dumpInternals(fd, args); 1525 dumpTracks(fd, args); 1526 dumpEffectChains(fd, args); 1527 return NO_ERROR; 1528} 1529 1530status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1531{ 1532 const size_t SIZE = 256; 1533 char buffer[SIZE]; 1534 String8 result; 1535 1536 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1537 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1538 const stream_type_t *st = &mStreamTypes[i]; 1539 if (i > 0) { 1540 result.appendFormat(", "); 1541 } 1542 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1543 if (st->mute) { 1544 result.append("M"); 1545 } 1546 } 1547 result.append("\n"); 1548 write(fd, result.string(), result.length()); 1549 result.clear(); 1550 1551 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1552 result.append(buffer); 1553 result.append(" Name Client Type Fmt Chn mask Session Frames S M F SRate L dB R dB " 1554 "Server User Main buf Aux Buf\n"); 1555 for (size_t i = 0; i < mTracks.size(); ++i) { 1556 sp<Track> track = mTracks[i]; 1557 if (track != 0) { 1558 track->dump(buffer, SIZE); 1559 result.append(buffer); 1560 } 1561 } 1562 1563 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1564 result.append(buffer); 1565 result.append(" Name Client Type Fmt Chn mask Session Frames S M F SRate L dB R dB " 1566 "Server User Main buf Aux Buf\n"); 1567 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1568 sp<Track> track = mActiveTracks[i].promote(); 1569 if (track != 0) { 1570 track->dump(buffer, SIZE); 1571 result.append(buffer); 1572 } 1573 } 1574 write(fd, result.string(), result.size()); 1575 return NO_ERROR; 1576} 1577 1578status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1579{ 1580 const size_t SIZE = 256; 1581 char buffer[SIZE]; 1582 String8 result; 1583 1584 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1585 result.append(buffer); 1586 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1587 result.append(buffer); 1588 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1589 result.append(buffer); 1590 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1591 result.append(buffer); 1592 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1593 result.append(buffer); 1594 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1595 result.append(buffer); 1596 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1597 result.append(buffer); 1598 write(fd, result.string(), result.size()); 1599 1600 dumpBase(fd, args); 1601 1602 return NO_ERROR; 1603} 1604 1605// Thread virtuals 1606status_t AudioFlinger::PlaybackThread::readyToRun() 1607{ 1608 status_t status = initCheck(); 1609 if (status == NO_ERROR) { 1610 ALOGI("AudioFlinger's thread %p ready to run", this); 1611 } else { 1612 ALOGE("No working audio driver found."); 1613 } 1614 return status; 1615} 1616 1617void AudioFlinger::PlaybackThread::onFirstRef() 1618{ 1619 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1620} 1621 1622// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1623sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1624 const sp<AudioFlinger::Client>& client, 1625 audio_stream_type_t streamType, 1626 uint32_t sampleRate, 1627 audio_format_t format, 1628 uint32_t channelMask, 1629 int frameCount, 1630 const sp<IMemory>& sharedBuffer, 1631 int sessionId, 1632 IAudioFlinger::track_flags_t flags, 1633 pid_t tid, 1634 status_t *status) 1635{ 1636 sp<Track> track; 1637 status_t lStatus; 1638 1639 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1640 1641 // client expresses a preference for FAST, but we get the final say 1642 if (flags & IAudioFlinger::TRACK_FAST) { 1643 if ( 1644 // not timed 1645 (!isTimed) && 1646 // either of these use cases: 1647 ( 1648 // use case 1: shared buffer with any frame count 1649 ( 1650 (sharedBuffer != 0) 1651 ) || 1652 // use case 2: callback handler and frame count is default or at least as large as HAL 1653 ( 1654 (tid != -1) && 1655 ((frameCount == 0) || 1656 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1657 ) 1658 ) && 1659 // PCM data 1660 audio_is_linear_pcm(format) && 1661 // mono or stereo 1662 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1663 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1664#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1665 // hardware sample rate 1666 (sampleRate == mSampleRate) && 1667#endif 1668 // normal mixer has an associated fast mixer 1669 hasFastMixer() && 1670 // there are sufficient fast track slots available 1671 (mFastTrackAvailMask != 0) 1672 // FIXME test that MixerThread for this fast track has a capable output HAL 1673 // FIXME add a permission test also? 1674 ) { 1675 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1676 if (frameCount == 0) { 1677 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1678 } 1679 ALOGI("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1680 frameCount, mFrameCount); 1681 } else { 1682 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1683 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1684 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1685 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1686 audio_is_linear_pcm(format), 1687 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1688 flags &= ~IAudioFlinger::TRACK_FAST; 1689 // For compatibility with AudioTrack calculation, buffer depth is forced 1690 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1691 // This is probably too conservative, but legacy application code may depend on it. 1692 // If you change this calculation, also review the start threshold which is related. 1693 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1694 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1695 if (minBufCount < 2) { 1696 minBufCount = 2; 1697 } 1698 int minFrameCount = mNormalFrameCount * minBufCount; 1699 if (frameCount < minFrameCount) { 1700 frameCount = minFrameCount; 1701 } 1702 } 1703 } 1704 1705 if (mType == DIRECT) { 1706 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1707 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1708 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1709 "for output %p with format %d", 1710 sampleRate, format, channelMask, mOutput, mFormat); 1711 lStatus = BAD_VALUE; 1712 goto Exit; 1713 } 1714 } 1715 } else { 1716 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1717 if (sampleRate > mSampleRate*2) { 1718 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1719 lStatus = BAD_VALUE; 1720 goto Exit; 1721 } 1722 } 1723 1724 lStatus = initCheck(); 1725 if (lStatus != NO_ERROR) { 1726 ALOGE("Audio driver not initialized."); 1727 goto Exit; 1728 } 1729 1730 { // scope for mLock 1731 Mutex::Autolock _l(mLock); 1732 1733 // all tracks in same audio session must share the same routing strategy otherwise 1734 // conflicts will happen when tracks are moved from one output to another by audio policy 1735 // manager 1736 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1737 for (size_t i = 0; i < mTracks.size(); ++i) { 1738 sp<Track> t = mTracks[i]; 1739 if (t != 0 && !t->isOutputTrack()) { 1740 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1741 if (sessionId == t->sessionId() && strategy != actual) { 1742 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1743 strategy, actual); 1744 lStatus = BAD_VALUE; 1745 goto Exit; 1746 } 1747 } 1748 } 1749 1750 if (!isTimed) { 1751 track = new Track(this, client, streamType, sampleRate, format, 1752 channelMask, frameCount, sharedBuffer, sessionId, flags); 1753 } else { 1754 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1755 channelMask, frameCount, sharedBuffer, sessionId); 1756 } 1757 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1758 lStatus = NO_MEMORY; 1759 goto Exit; 1760 } 1761 mTracks.add(track); 1762 1763 sp<EffectChain> chain = getEffectChain_l(sessionId); 1764 if (chain != 0) { 1765 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1766 track->setMainBuffer(chain->inBuffer()); 1767 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1768 chain->incTrackCnt(); 1769 } 1770 } 1771 1772#ifdef HAVE_REQUEST_PRIORITY 1773 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1774 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1775 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1776 // so ask activity manager to do this on our behalf 1777 int err = requestPriority(callingPid, tid, 1); 1778 if (err != 0) { 1779 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1780 1, callingPid, tid, err); 1781 } 1782 } 1783#endif 1784 1785 lStatus = NO_ERROR; 1786 1787Exit: 1788 if (status) { 1789 *status = lStatus; 1790 } 1791 return track; 1792} 1793 1794uint32_t AudioFlinger::PlaybackThread::latency() const 1795{ 1796 Mutex::Autolock _l(mLock); 1797 if (initCheck() == NO_ERROR) { 1798 return mOutput->stream->get_latency(mOutput->stream); 1799 } else { 1800 return 0; 1801 } 1802} 1803 1804void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1805{ 1806 Mutex::Autolock _l(mLock); 1807 mMasterVolume = value; 1808} 1809 1810void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1811{ 1812 Mutex::Autolock _l(mLock); 1813 setMasterMute_l(muted); 1814} 1815 1816void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1817{ 1818 Mutex::Autolock _l(mLock); 1819 mStreamTypes[stream].volume = value; 1820} 1821 1822void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1823{ 1824 Mutex::Autolock _l(mLock); 1825 mStreamTypes[stream].mute = muted; 1826} 1827 1828float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1829{ 1830 Mutex::Autolock _l(mLock); 1831 return mStreamTypes[stream].volume; 1832} 1833 1834// addTrack_l() must be called with ThreadBase::mLock held 1835status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1836{ 1837 status_t status = ALREADY_EXISTS; 1838 1839 // set retry count for buffer fill 1840 track->mRetryCount = kMaxTrackStartupRetries; 1841 if (mActiveTracks.indexOf(track) < 0) { 1842 // the track is newly added, make sure it fills up all its 1843 // buffers before playing. This is to ensure the client will 1844 // effectively get the latency it requested. 1845 track->mFillingUpStatus = Track::FS_FILLING; 1846 track->mResetDone = false; 1847 mActiveTracks.add(track); 1848 if (track->mainBuffer() != mMixBuffer) { 1849 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1850 if (chain != 0) { 1851 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1852 chain->incActiveTrackCnt(); 1853 } 1854 } 1855 1856 status = NO_ERROR; 1857 } 1858 1859 ALOGV("mWaitWorkCV.broadcast"); 1860 mWaitWorkCV.broadcast(); 1861 1862 return status; 1863} 1864 1865// destroyTrack_l() must be called with ThreadBase::mLock held 1866void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1867{ 1868 track->mState = TrackBase::TERMINATED; 1869 if (mActiveTracks.indexOf(track) < 0) { 1870 removeTrack_l(track); 1871 } 1872} 1873 1874void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1875{ 1876 mTracks.remove(track); 1877 deleteTrackName_l(track->name()); 1878 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1879 if (chain != 0) { 1880 chain->decTrackCnt(); 1881 } 1882} 1883 1884String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1885{ 1886 String8 out_s8 = String8(""); 1887 char *s; 1888 1889 Mutex::Autolock _l(mLock); 1890 if (initCheck() != NO_ERROR) { 1891 return out_s8; 1892 } 1893 1894 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1895 out_s8 = String8(s); 1896 free(s); 1897 return out_s8; 1898} 1899 1900// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1901void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1902 AudioSystem::OutputDescriptor desc; 1903 void *param2 = NULL; 1904 1905 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1906 1907 switch (event) { 1908 case AudioSystem::OUTPUT_OPENED: 1909 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1910 desc.channels = mChannelMask; 1911 desc.samplingRate = mSampleRate; 1912 desc.format = mFormat; 1913 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1914 desc.latency = latency(); 1915 param2 = &desc; 1916 break; 1917 1918 case AudioSystem::STREAM_CONFIG_CHANGED: 1919 param2 = ¶m; 1920 case AudioSystem::OUTPUT_CLOSED: 1921 default: 1922 break; 1923 } 1924 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1925} 1926 1927void AudioFlinger::PlaybackThread::readOutputParameters() 1928{ 1929 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1930 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1931 mChannelCount = (uint16_t)popcount(mChannelMask); 1932 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1933 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1934 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1935 if (mFrameCount & 15) { 1936 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1937 mFrameCount); 1938 } 1939 1940 // Calculate size of normal mix buffer relative to the HAL output buffer size 1941 uint32_t multiple = 1; 1942 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1943 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1944 multiple = (minNormalFrameCount + mFrameCount - 1) / mFrameCount; 1945 // force multiple to be even, for compatibility with doubling of fast tracks due to HAL SRC 1946 // (it would be unusual for the normal mix buffer size to not be a multiple of fast track) 1947 // FIXME this rounding up should not be done if no HAL SRC 1948 if ((multiple > 2) && (multiple & 1)) { 1949 ++multiple; 1950 } 1951 } 1952 mNormalFrameCount = multiple * mFrameCount; 1953 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 1954 1955 // FIXME - Current mixer implementation only supports stereo output: Always 1956 // Allocate a stereo buffer even if HW output is mono. 1957 delete[] mMixBuffer; 1958 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 1959 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 1960 1961 // force reconfiguration of effect chains and engines to take new buffer size and audio 1962 // parameters into account 1963 // Note that mLock is not held when readOutputParameters() is called from the constructor 1964 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1965 // matter. 1966 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1967 Vector< sp<EffectChain> > effectChains = mEffectChains; 1968 for (size_t i = 0; i < effectChains.size(); i ++) { 1969 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1970 } 1971} 1972 1973status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1974{ 1975 if (halFrames == NULL || dspFrames == NULL) { 1976 return BAD_VALUE; 1977 } 1978 Mutex::Autolock _l(mLock); 1979 if (initCheck() != NO_ERROR) { 1980 return INVALID_OPERATION; 1981 } 1982 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1983 1984 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1985} 1986 1987uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1988{ 1989 Mutex::Autolock _l(mLock); 1990 uint32_t result = 0; 1991 if (getEffectChain_l(sessionId) != 0) { 1992 result = EFFECT_SESSION; 1993 } 1994 1995 for (size_t i = 0; i < mTracks.size(); ++i) { 1996 sp<Track> track = mTracks[i]; 1997 if (sessionId == track->sessionId() && 1998 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1999 result |= TRACK_SESSION; 2000 break; 2001 } 2002 } 2003 2004 return result; 2005} 2006 2007uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2008{ 2009 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2010 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2011 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2012 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2013 } 2014 for (size_t i = 0; i < mTracks.size(); i++) { 2015 sp<Track> track = mTracks[i]; 2016 if (sessionId == track->sessionId() && 2017 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2018 return AudioSystem::getStrategyForStream(track->streamType()); 2019 } 2020 } 2021 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2022} 2023 2024 2025AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2026{ 2027 Mutex::Autolock _l(mLock); 2028 return mOutput; 2029} 2030 2031AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2032{ 2033 Mutex::Autolock _l(mLock); 2034 AudioStreamOut *output = mOutput; 2035 mOutput = NULL; 2036 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2037 // must push a NULL and wait for ack 2038 mOutputSink.clear(); 2039 mPipeSink.clear(); 2040 mNormalSink.clear(); 2041 return output; 2042} 2043 2044// this method must always be called either with ThreadBase mLock held or inside the thread loop 2045audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2046{ 2047 if (mOutput == NULL) { 2048 return NULL; 2049 } 2050 return &mOutput->stream->common; 2051} 2052 2053uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2054{ 2055 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2056 // decoding and transfer time. So sleeping for half of the latency would likely cause 2057 // underruns 2058 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2059 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2060 } else { 2061 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2062 } 2063} 2064 2065status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2066{ 2067 if (!isValidSyncEvent(event)) { 2068 return BAD_VALUE; 2069 } 2070 2071 Mutex::Autolock _l(mLock); 2072 2073 for (size_t i = 0; i < mTracks.size(); ++i) { 2074 sp<Track> track = mTracks[i]; 2075 if (event->triggerSession() == track->sessionId()) { 2076 track->setSyncEvent(event); 2077 return NO_ERROR; 2078 } 2079 } 2080 2081 return NAME_NOT_FOUND; 2082} 2083 2084bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2085{ 2086 switch (event->type()) { 2087 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2088 return true; 2089 default: 2090 break; 2091 } 2092 return false; 2093} 2094 2095// ---------------------------------------------------------------------------- 2096 2097AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2098 audio_io_handle_t id, uint32_t device, type_t type) 2099 : PlaybackThread(audioFlinger, output, id, device, type), 2100 // mAudioMixer below 2101#ifdef SOAKER 2102 mSoaker(NULL), 2103#endif 2104 // mFastMixer below 2105 mFastMixerFutex(0) 2106 // mOutputSink below 2107 // mPipeSink below 2108 // mNormalSink below 2109{ 2110 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2111 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2112 "mFrameCount=%d, mNormalFrameCount=%d", 2113 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2114 mNormalFrameCount); 2115 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2116 2117 // FIXME - Current mixer implementation only supports stereo output 2118 if (mChannelCount == 1) { 2119 ALOGE("Invalid audio hardware channel count"); 2120 } 2121 2122 // create an NBAIO sink for the HAL output stream, and negotiate 2123 mOutputSink = new AudioStreamOutSink(output->stream); 2124 size_t numCounterOffers = 0; 2125 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2126 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2127 ALOG_ASSERT(index == 0); 2128 2129 // initialize fast mixer depending on configuration 2130 bool initFastMixer; 2131 switch (kUseFastMixer) { 2132 case FastMixer_Never: 2133 initFastMixer = false; 2134 break; 2135 case FastMixer_Always: 2136 initFastMixer = true; 2137 break; 2138 case FastMixer_Static: 2139 case FastMixer_Dynamic: 2140 initFastMixer = mFrameCount < mNormalFrameCount; 2141 break; 2142 } 2143 if (initFastMixer) { 2144 2145 // create a MonoPipe to connect our submix to FastMixer 2146 NBAIO_Format format = mOutputSink->format(); 2147 // frame count will be rounded up to a power of 2, so this formula should work well 2148 MonoPipe *monoPipe = new MonoPipe((mNormalFrameCount * 3) / 2, format, 2149 true /*writeCanBlock*/); 2150 const NBAIO_Format offers[1] = {format}; 2151 size_t numCounterOffers = 0; 2152 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2153 ALOG_ASSERT(index == 0); 2154 mPipeSink = monoPipe; 2155 2156#ifdef SOAKER 2157 // create a soaker as workaround for governor issues 2158 mSoaker = new Soaker(); 2159 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2160 mSoaker->run("Soaker", PRIORITY_LOWEST); 2161#endif 2162 2163 // create fast mixer and configure it initially with just one fast track for our submix 2164 mFastMixer = new FastMixer(); 2165 FastMixerStateQueue *sq = mFastMixer->sq(); 2166 FastMixerState *state = sq->begin(); 2167 FastTrack *fastTrack = &state->mFastTracks[0]; 2168 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2169 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2170 fastTrack->mVolumeProvider = NULL; 2171 fastTrack->mGeneration++; 2172 state->mFastTracksGen++; 2173 state->mTrackMask = 1; 2174 // fast mixer will use the HAL output sink 2175 state->mOutputSink = mOutputSink.get(); 2176 state->mOutputSinkGen++; 2177 state->mFrameCount = mFrameCount; 2178 state->mCommand = FastMixerState::COLD_IDLE; 2179 // already done in constructor initialization list 2180 //mFastMixerFutex = 0; 2181 state->mColdFutexAddr = &mFastMixerFutex; 2182 state->mColdGen++; 2183 state->mDumpState = &mFastMixerDumpState; 2184 sq->end(); 2185 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2186 2187 // start the fast mixer 2188 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2189#ifdef HAVE_REQUEST_PRIORITY 2190 pid_t tid = mFastMixer->getTid(); 2191 int err = requestPriority(getpid_cached, tid, 2); 2192 if (err != 0) { 2193 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2194 2, getpid_cached, tid, err); 2195 } 2196#endif 2197 2198 } else { 2199 mFastMixer = NULL; 2200 } 2201 2202 switch (kUseFastMixer) { 2203 case FastMixer_Never: 2204 case FastMixer_Dynamic: 2205 mNormalSink = mOutputSink; 2206 break; 2207 case FastMixer_Always: 2208 mNormalSink = mPipeSink; 2209 break; 2210 case FastMixer_Static: 2211 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2212 break; 2213 } 2214} 2215 2216AudioFlinger::MixerThread::~MixerThread() 2217{ 2218 if (mFastMixer != NULL) { 2219 FastMixerStateQueue *sq = mFastMixer->sq(); 2220 FastMixerState *state = sq->begin(); 2221 if (state->mCommand == FastMixerState::COLD_IDLE) { 2222 int32_t old = android_atomic_inc(&mFastMixerFutex); 2223 if (old == -1) { 2224 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2225 } 2226 } 2227 state->mCommand = FastMixerState::EXIT; 2228 sq->end(); 2229 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2230 mFastMixer->join(); 2231 // Though the fast mixer thread has exited, it's state queue is still valid. 2232 // We'll use that extract the final state which contains one remaining fast track 2233 // corresponding to our sub-mix. 2234 state = sq->begin(); 2235 ALOG_ASSERT(state->mTrackMask == 1); 2236 FastTrack *fastTrack = &state->mFastTracks[0]; 2237 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2238 delete fastTrack->mBufferProvider; 2239 sq->end(false /*didModify*/); 2240 delete mFastMixer; 2241#ifdef SOAKER 2242 if (mSoaker != NULL) { 2243 mSoaker->requestExitAndWait(); 2244 } 2245 delete mSoaker; 2246#endif 2247 } 2248 delete mAudioMixer; 2249} 2250 2251class CpuStats { 2252public: 2253 CpuStats(); 2254 void sample(const String8 &title); 2255#ifdef DEBUG_CPU_USAGE 2256private: 2257 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2258 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2259 2260 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2261 2262 int mCpuNum; // thread's current CPU number 2263 int mCpukHz; // frequency of thread's current CPU in kHz 2264#endif 2265}; 2266 2267CpuStats::CpuStats() 2268#ifdef DEBUG_CPU_USAGE 2269 : mCpuNum(-1), mCpukHz(-1) 2270#endif 2271{ 2272} 2273 2274void CpuStats::sample(const String8 &title) { 2275#ifdef DEBUG_CPU_USAGE 2276 // get current thread's delta CPU time in wall clock ns 2277 double wcNs; 2278 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2279 2280 // record sample for wall clock statistics 2281 if (valid) { 2282 mWcStats.sample(wcNs); 2283 } 2284 2285 // get the current CPU number 2286 int cpuNum = sched_getcpu(); 2287 2288 // get the current CPU frequency in kHz 2289 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2290 2291 // check if either CPU number or frequency changed 2292 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2293 mCpuNum = cpuNum; 2294 mCpukHz = cpukHz; 2295 // ignore sample for purposes of cycles 2296 valid = false; 2297 } 2298 2299 // if no change in CPU number or frequency, then record sample for cycle statistics 2300 if (valid && mCpukHz > 0) { 2301 double cycles = wcNs * cpukHz * 0.000001; 2302 mHzStats.sample(cycles); 2303 } 2304 2305 unsigned n = mWcStats.n(); 2306 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2307 if ((n & 127) == 1) { 2308 long long elapsed = mCpuUsage.elapsed(); 2309 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2310 double perLoop = elapsed / (double) n; 2311 double perLoop100 = perLoop * 0.01; 2312 double perLoop1k = perLoop * 0.001; 2313 double mean = mWcStats.mean(); 2314 double stddev = mWcStats.stddev(); 2315 double minimum = mWcStats.minimum(); 2316 double maximum = mWcStats.maximum(); 2317 double meanCycles = mHzStats.mean(); 2318 double stddevCycles = mHzStats.stddev(); 2319 double minCycles = mHzStats.minimum(); 2320 double maxCycles = mHzStats.maximum(); 2321 mCpuUsage.resetElapsed(); 2322 mWcStats.reset(); 2323 mHzStats.reset(); 2324 ALOGD("CPU usage for %s over past %.1f secs\n" 2325 " (%u mixer loops at %.1f mean ms per loop):\n" 2326 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2327 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2328 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2329 title.string(), 2330 elapsed * .000000001, n, perLoop * .000001, 2331 mean * .001, 2332 stddev * .001, 2333 minimum * .001, 2334 maximum * .001, 2335 mean / perLoop100, 2336 stddev / perLoop100, 2337 minimum / perLoop100, 2338 maximum / perLoop100, 2339 meanCycles / perLoop1k, 2340 stddevCycles / perLoop1k, 2341 minCycles / perLoop1k, 2342 maxCycles / perLoop1k); 2343 2344 } 2345 } 2346#endif 2347}; 2348 2349void AudioFlinger::PlaybackThread::checkSilentMode_l() 2350{ 2351 if (!mMasterMute) { 2352 char value[PROPERTY_VALUE_MAX]; 2353 if (property_get("ro.audio.silent", value, "0") > 0) { 2354 char *endptr; 2355 unsigned long ul = strtoul(value, &endptr, 0); 2356 if (*endptr == '\0' && ul != 0) { 2357 ALOGD("Silence is golden"); 2358 // The setprop command will not allow a property to be changed after 2359 // the first time it is set, so we don't have to worry about un-muting. 2360 setMasterMute_l(true); 2361 } 2362 } 2363 } 2364} 2365 2366bool AudioFlinger::PlaybackThread::threadLoop() 2367{ 2368 Vector< sp<Track> > tracksToRemove; 2369 2370 standbyTime = systemTime(); 2371 2372 // MIXER 2373 nsecs_t lastWarning = 0; 2374if (mType == MIXER) { 2375 longStandbyExit = false; 2376} 2377 2378 // DUPLICATING 2379 // FIXME could this be made local to while loop? 2380 writeFrames = 0; 2381 2382 cacheParameters_l(); 2383 sleepTime = idleSleepTime; 2384 2385if (mType == MIXER) { 2386 sleepTimeShift = 0; 2387} 2388 2389 CpuStats cpuStats; 2390 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2391 2392 acquireWakeLock(); 2393 2394 while (!exitPending()) 2395 { 2396 cpuStats.sample(myName); 2397 2398 Vector< sp<EffectChain> > effectChains; 2399 2400 processConfigEvents(); 2401 2402 { // scope for mLock 2403 2404 Mutex::Autolock _l(mLock); 2405 2406 if (checkForNewParameters_l()) { 2407 cacheParameters_l(); 2408 } 2409 2410 saveOutputTracks(); 2411 2412 // put audio hardware into standby after short delay 2413 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2414 mSuspended > 0)) { 2415 if (!mStandby) { 2416 2417 threadLoop_standby(); 2418 2419 mStandby = true; 2420 mBytesWritten = 0; 2421 } 2422 2423 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2424 // we're about to wait, flush the binder command buffer 2425 IPCThreadState::self()->flushCommands(); 2426 2427 clearOutputTracks(); 2428 2429 if (exitPending()) break; 2430 2431 releaseWakeLock_l(); 2432 // wait until we have something to do... 2433 ALOGV("%s going to sleep", myName.string()); 2434 mWaitWorkCV.wait(mLock); 2435 ALOGV("%s waking up", myName.string()); 2436 acquireWakeLock_l(); 2437 2438 mMixerStatus = MIXER_IDLE; 2439 2440 checkSilentMode_l(); 2441 2442 standbyTime = systemTime() + standbyDelay; 2443 sleepTime = idleSleepTime; 2444 if (mType == MIXER) { 2445 sleepTimeShift = 0; 2446 } 2447 2448 continue; 2449 } 2450 } 2451 2452 mMixerStatus = prepareTracks_l(&tracksToRemove); 2453 2454 // prevent any changes in effect chain list and in each effect chain 2455 // during mixing and effect process as the audio buffers could be deleted 2456 // or modified if an effect is created or deleted 2457 lockEffectChains_l(effectChains); 2458 } 2459 2460 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2461 threadLoop_mix(); 2462 } else { 2463 threadLoop_sleepTime(); 2464 } 2465 2466 if (mSuspended > 0) { 2467 sleepTime = suspendSleepTimeUs(); 2468 } 2469 2470 // only process effects if we're going to write 2471 if (sleepTime == 0) { 2472 for (size_t i = 0; i < effectChains.size(); i ++) { 2473 effectChains[i]->process_l(); 2474 } 2475 } 2476 2477 // enable changes in effect chain 2478 unlockEffectChains(effectChains); 2479 2480 // sleepTime == 0 means we must write to audio hardware 2481 if (sleepTime == 0) { 2482 2483 threadLoop_write(); 2484 2485if (mType == MIXER) { 2486 // write blocked detection 2487 nsecs_t now = systemTime(); 2488 nsecs_t delta = now - mLastWriteTime; 2489 if (!mStandby && delta > maxPeriod) { 2490 mNumDelayedWrites++; 2491 if ((now - lastWarning) > kWarningThrottleNs) { 2492 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2493 ns2ms(delta), mNumDelayedWrites, this); 2494 lastWarning = now; 2495 } 2496 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2497 // a different threshold. Or completely removed for what it is worth anyway... 2498 if (mStandby) { 2499 longStandbyExit = true; 2500 } 2501 } 2502} 2503 2504 mStandby = false; 2505 } else { 2506 usleep(sleepTime); 2507 } 2508 2509 // Finally let go of removed track(s), without the lock held 2510 // since we can't guarantee the destructors won't acquire that 2511 // same lock. This will also mutate and push a new fast mixer state. 2512 threadLoop_removeTracks(tracksToRemove); 2513 tracksToRemove.clear(); 2514 2515 // FIXME I don't understand the need for this here; 2516 // it was in the original code but maybe the 2517 // assignment in saveOutputTracks() makes this unnecessary? 2518 clearOutputTracks(); 2519 2520 // Effect chains will be actually deleted here if they were removed from 2521 // mEffectChains list during mixing or effects processing 2522 effectChains.clear(); 2523 2524 // FIXME Note that the above .clear() is no longer necessary since effectChains 2525 // is now local to this block, but will keep it for now (at least until merge done). 2526 } 2527 2528if (mType == MIXER || mType == DIRECT) { 2529 // put output stream into standby mode 2530 if (!mStandby) { 2531 mOutput->stream->common.standby(&mOutput->stream->common); 2532 } 2533} 2534if (mType == DUPLICATING) { 2535 // for DuplicatingThread, standby mode is handled by the outputTracks 2536} 2537 2538 releaseWakeLock(); 2539 2540 ALOGV("Thread %p type %d exiting", this, mType); 2541 return false; 2542} 2543 2544// FIXME This method needs a better name. 2545// It pushes a new fast mixer state and returns (via tracksToRemove) a set of tracks to remove. 2546void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2547{ 2548 // were any of the removed tracks also fast tracks? 2549 unsigned removedMask = 0; 2550 for (size_t i = 0; i < tracksToRemove.size(); ++i) { 2551 if (tracksToRemove[i]->isFastTrack()) { 2552 int j = tracksToRemove[i]->mFastIndex; 2553 ALOG_ASSERT(0 < j && j < FastMixerState::kMaxFastTracks); 2554 removedMask |= 1 << j; 2555 } 2556 } 2557 Track* newArray[FastMixerState::kMaxFastTracks]; 2558 unsigned newMask; 2559 { 2560 AutoMutex _l(mLock); 2561 mFastTrackAvailMask |= removedMask; 2562 newMask = mFastTrackNewMask; 2563 if (newMask) { 2564 mFastTrackNewMask = 0; 2565 memcpy(newArray, mFastTrackNewArray, sizeof(mFastTrackNewArray)); 2566#if !LOG_NDEBUG 2567 memset(mFastTrackNewArray, 0, sizeof(mFastTrackNewArray)); 2568#endif 2569 } 2570 } 2571 unsigned changedMask = newMask | removedMask; 2572 // are there any newly added or removed fast tracks? 2573 if (changedMask) { 2574 2575 // This assert would be incorrect because it's theoretically possible (though unlikely) 2576 // for a track to be created and then removed within the same normal mix cycle: 2577 // ALOG_ASSERT(!(newMask & removedMask)); 2578 // The converse, of removing a track and then creating a new track at the identical slot 2579 // within the same normal mix cycle, is impossible because the slot isn't marked available. 2580 2581 // prepare a new state to push 2582 FastMixerStateQueue *sq = mFastMixer->sq(); 2583 FastMixerState *state = sq->begin(); 2584 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2585 while (changedMask) { 2586 int j = __builtin_ctz(changedMask); 2587 ALOG_ASSERT(0 < j && j < FastMixerState::kMaxFastTracks); 2588 changedMask &= ~(1 << j); 2589 FastTrack *fastTrack = &state->mFastTracks[j]; 2590 // must first do new tracks, then removed tracks, in case same track in both 2591 if (newMask & (1 << j)) { 2592 ALOG_ASSERT(!(state->mTrackMask & (1 << j))); 2593 ALOG_ASSERT(fastTrack->mBufferProvider == NULL && 2594 fastTrack->mVolumeProvider == NULL); 2595 Track *track = newArray[j]; 2596 AudioBufferProvider *abp = track; 2597 VolumeProvider *vp = track; 2598 fastTrack->mBufferProvider = abp; 2599 fastTrack->mVolumeProvider = vp; 2600 fastTrack->mSampleRate = track->mSampleRate; 2601 fastTrack->mChannelMask = track->mChannelMask; 2602 state->mTrackMask |= 1 << j; 2603 } 2604 if (removedMask & (1 << j)) { 2605 ALOG_ASSERT(state->mTrackMask & (1 << j)); 2606 ALOG_ASSERT(fastTrack->mBufferProvider != NULL && 2607 fastTrack->mVolumeProvider != NULL); 2608 fastTrack->mBufferProvider = NULL; 2609 fastTrack->mVolumeProvider = NULL; 2610 fastTrack->mSampleRate = mSampleRate; 2611 fastTrack->mChannelMask = AUDIO_CHANNEL_OUT_STEREO; 2612 state->mTrackMask &= ~(1 << j); 2613 } 2614 fastTrack->mGeneration++; 2615 } 2616 state->mFastTracksGen++; 2617 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2618 if (kUseFastMixer == FastMixer_Dynamic && 2619 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2620 state->mCommand = FastMixerState::COLD_IDLE; 2621 state->mColdFutexAddr = &mFastMixerFutex; 2622 state->mColdGen++; 2623 mFastMixerFutex = 0; 2624 mNormalSink = mOutputSink; 2625 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2626 } 2627 sq->end(); 2628 // If any fast tracks were removed, we must wait for acknowledgement 2629 // because we're about to decrement the last sp<> on those tracks. 2630 // Similarly if we put it into cold idle, need to wait for acknowledgement 2631 // so that it stops doing I/O. 2632 if (removedMask) { 2633 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2634 } 2635 sq->push(block); 2636 } 2637 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2638} 2639 2640void AudioFlinger::MixerThread::threadLoop_write() 2641{ 2642 // FIXME we should only do one push per cycle; confirm this is true 2643 // Start the fast mixer if it's not already running 2644 if (mFastMixer != NULL) { 2645 FastMixerStateQueue *sq = mFastMixer->sq(); 2646 FastMixerState *state = sq->begin(); 2647 if (state->mCommand != FastMixerState::MIX_WRITE && 2648 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2649 if (state->mCommand == FastMixerState::COLD_IDLE) { 2650 int32_t old = android_atomic_inc(&mFastMixerFutex); 2651 if (old == -1) { 2652 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2653 } 2654 } 2655 state->mCommand = FastMixerState::MIX_WRITE; 2656 sq->end(); 2657 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2658 if (kUseFastMixer == FastMixer_Dynamic) { 2659 mNormalSink = mPipeSink; 2660 } 2661 } else { 2662 sq->end(false /*didModify*/); 2663 } 2664 } 2665 PlaybackThread::threadLoop_write(); 2666} 2667 2668// shared by MIXER and DIRECT, overridden by DUPLICATING 2669void AudioFlinger::PlaybackThread::threadLoop_write() 2670{ 2671 // FIXME rewrite to reduce number of system calls 2672 mLastWriteTime = systemTime(); 2673 mInWrite = true; 2674 2675#define mBitShift 2 // FIXME 2676 size_t count = mixBufferSize >> mBitShift; 2677 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2678 if (framesWritten > 0) { 2679 size_t bytesWritten = framesWritten << mBitShift; 2680 mBytesWritten += bytesWritten; 2681 } 2682 2683 mNumWrites++; 2684 mInWrite = false; 2685} 2686 2687void AudioFlinger::MixerThread::threadLoop_standby() 2688{ 2689 // Idle the fast mixer if it's currently running 2690 if (mFastMixer != NULL) { 2691 FastMixerStateQueue *sq = mFastMixer->sq(); 2692 FastMixerState *state = sq->begin(); 2693 if (!(state->mCommand & FastMixerState::IDLE)) { 2694 state->mCommand = FastMixerState::COLD_IDLE; 2695 state->mColdFutexAddr = &mFastMixerFutex; 2696 state->mColdGen++; 2697 mFastMixerFutex = 0; 2698 sq->end(); 2699 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2700 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2701 if (kUseFastMixer == FastMixer_Dynamic) { 2702 mNormalSink = mOutputSink; 2703 } 2704 } else { 2705 sq->end(false /*didModify*/); 2706 } 2707 } 2708 PlaybackThread::threadLoop_standby(); 2709} 2710 2711// shared by MIXER and DIRECT, overridden by DUPLICATING 2712void AudioFlinger::PlaybackThread::threadLoop_standby() 2713{ 2714 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2715 mOutput->stream->common.standby(&mOutput->stream->common); 2716} 2717 2718void AudioFlinger::MixerThread::threadLoop_mix() 2719{ 2720 // obtain the presentation timestamp of the next output buffer 2721 int64_t pts; 2722 status_t status = INVALID_OPERATION; 2723 2724 if (NULL != mOutput->stream->get_next_write_timestamp) { 2725 status = mOutput->stream->get_next_write_timestamp( 2726 mOutput->stream, &pts); 2727 } 2728 2729 if (status != NO_ERROR) { 2730 pts = AudioBufferProvider::kInvalidPTS; 2731 } 2732 2733 // mix buffers... 2734 mAudioMixer->process(pts); 2735 // increase sleep time progressively when application underrun condition clears. 2736 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2737 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2738 // such that we would underrun the audio HAL. 2739 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2740 sleepTimeShift--; 2741 } 2742 sleepTime = 0; 2743 standbyTime = systemTime() + standbyDelay; 2744 //TODO: delay standby when effects have a tail 2745} 2746 2747void AudioFlinger::MixerThread::threadLoop_sleepTime() 2748{ 2749 // If no tracks are ready, sleep once for the duration of an output 2750 // buffer size, then write 0s to the output 2751 if (sleepTime == 0) { 2752 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2753 sleepTime = activeSleepTime >> sleepTimeShift; 2754 if (sleepTime < kMinThreadSleepTimeUs) { 2755 sleepTime = kMinThreadSleepTimeUs; 2756 } 2757 // reduce sleep time in case of consecutive application underruns to avoid 2758 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2759 // duration we would end up writing less data than needed by the audio HAL if 2760 // the condition persists. 2761 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2762 sleepTimeShift++; 2763 } 2764 } else { 2765 sleepTime = idleSleepTime; 2766 } 2767 } else if (mBytesWritten != 0 || 2768 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2769 memset (mMixBuffer, 0, mixBufferSize); 2770 sleepTime = 0; 2771 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2772 } 2773 // TODO add standby time extension fct of effect tail 2774} 2775 2776// prepareTracks_l() must be called with ThreadBase::mLock held 2777AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2778 Vector< sp<Track> > *tracksToRemove) 2779{ 2780 2781 mixer_state mixerStatus = MIXER_IDLE; 2782 // find out which tracks need to be processed 2783 size_t count = mActiveTracks.size(); 2784 size_t mixedTracks = 0; 2785 size_t tracksWithEffect = 0; 2786 size_t fastTracks = 0; 2787 2788 float masterVolume = mMasterVolume; 2789 bool masterMute = mMasterMute; 2790 2791 if (masterMute) { 2792 masterVolume = 0; 2793 } 2794 // Delegate master volume control to effect in output mix effect chain if needed 2795 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2796 if (chain != 0) { 2797 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2798 chain->setVolume_l(&v, &v); 2799 masterVolume = (float)((v + (1 << 23)) >> 24); 2800 chain.clear(); 2801 } 2802 2803 for (size_t i=0 ; i<count ; i++) { 2804 sp<Track> t = mActiveTracks[i].promote(); 2805 if (t == 0) continue; 2806 2807 // this const just means the local variable doesn't change 2808 Track* const track = t.get(); 2809 2810 if (track->isFastTrack()) { 2811 // cache the combined master volume and stream type volume for fast mixer; 2812 // this lacks any synchronization or barrier so VolumeProvider may read a stale value 2813 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2814 ++fastTracks; 2815 if (track->isTerminated()) { 2816 tracksToRemove->add(track); 2817 } 2818 continue; 2819 } 2820 2821 { // local variable scope to avoid goto warning 2822 2823 audio_track_cblk_t* cblk = track->cblk(); 2824 2825 // The first time a track is added we wait 2826 // for all its buffers to be filled before processing it 2827 int name = track->name(); 2828 // make sure that we have enough frames to mix one full buffer. 2829 // enforce this condition only once to enable draining the buffer in case the client 2830 // app does not call stop() and relies on underrun to stop: 2831 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2832 // during last round 2833 uint32_t minFrames = 1; 2834 if (!track->isStopped() && !track->isPausing() && 2835 (mMixerStatus == MIXER_TRACKS_READY)) { 2836 if (t->sampleRate() == (int)mSampleRate) { 2837 minFrames = mNormalFrameCount; 2838 } else { 2839 // +1 for rounding and +1 for additional sample needed for interpolation 2840 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2841 // add frames already consumed but not yet released by the resampler 2842 // because cblk->framesReady() will include these frames 2843 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2844 // the minimum track buffer size is normally twice the number of frames necessary 2845 // to fill one buffer and the resampler should not leave more than one buffer worth 2846 // of unreleased frames after each pass, but just in case... 2847 ALOG_ASSERT(minFrames <= cblk->frameCount); 2848 } 2849 } 2850 if ((track->framesReady() >= minFrames) && track->isReady() && 2851 !track->isPaused() && !track->isTerminated()) 2852 { 2853 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2854 2855 mixedTracks++; 2856 2857 // track->mainBuffer() != mMixBuffer means there is an effect chain 2858 // connected to the track 2859 chain.clear(); 2860 if (track->mainBuffer() != mMixBuffer) { 2861 chain = getEffectChain_l(track->sessionId()); 2862 // Delegate volume control to effect in track effect chain if needed 2863 if (chain != 0) { 2864 tracksWithEffect++; 2865 } else { 2866 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2867 name, track->sessionId()); 2868 } 2869 } 2870 2871 2872 int param = AudioMixer::VOLUME; 2873 if (track->mFillingUpStatus == Track::FS_FILLED) { 2874 // no ramp for the first volume setting 2875 track->mFillingUpStatus = Track::FS_ACTIVE; 2876 if (track->mState == TrackBase::RESUMING) { 2877 track->mState = TrackBase::ACTIVE; 2878 param = AudioMixer::RAMP_VOLUME; 2879 } 2880 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2881 } else if (cblk->server != 0) { 2882 // If the track is stopped before the first frame was mixed, 2883 // do not apply ramp 2884 param = AudioMixer::RAMP_VOLUME; 2885 } 2886 2887 // compute volume for this track 2888 uint32_t vl, vr, va; 2889 if (track->isMuted() || track->isPausing() || 2890 mStreamTypes[track->streamType()].mute) { 2891 vl = vr = va = 0; 2892 if (track->isPausing()) { 2893 track->setPaused(); 2894 } 2895 } else { 2896 2897 // read original volumes with volume control 2898 float typeVolume = mStreamTypes[track->streamType()].volume; 2899 float v = masterVolume * typeVolume; 2900 uint32_t vlr = cblk->getVolumeLR(); 2901 vl = vlr & 0xFFFF; 2902 vr = vlr >> 16; 2903 // track volumes come from shared memory, so can't be trusted and must be clamped 2904 if (vl > MAX_GAIN_INT) { 2905 ALOGV("Track left volume out of range: %04X", vl); 2906 vl = MAX_GAIN_INT; 2907 } 2908 if (vr > MAX_GAIN_INT) { 2909 ALOGV("Track right volume out of range: %04X", vr); 2910 vr = MAX_GAIN_INT; 2911 } 2912 // now apply the master volume and stream type volume 2913 vl = (uint32_t)(v * vl) << 12; 2914 vr = (uint32_t)(v * vr) << 12; 2915 // assuming master volume and stream type volume each go up to 1.0, 2916 // vl and vr are now in 8.24 format 2917 2918 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2919 // send level comes from shared memory and so may be corrupt 2920 if (sendLevel > MAX_GAIN_INT) { 2921 ALOGV("Track send level out of range: %04X", sendLevel); 2922 sendLevel = MAX_GAIN_INT; 2923 } 2924 va = (uint32_t)(v * sendLevel); 2925 } 2926 // Delegate volume control to effect in track effect chain if needed 2927 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2928 // Do not ramp volume if volume is controlled by effect 2929 param = AudioMixer::VOLUME; 2930 track->mHasVolumeController = true; 2931 } else { 2932 // force no volume ramp when volume controller was just disabled or removed 2933 // from effect chain to avoid volume spike 2934 if (track->mHasVolumeController) { 2935 param = AudioMixer::VOLUME; 2936 } 2937 track->mHasVolumeController = false; 2938 } 2939 2940 // Convert volumes from 8.24 to 4.12 format 2941 // This additional clamping is needed in case chain->setVolume_l() overshot 2942 vl = (vl + (1 << 11)) >> 12; 2943 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2944 vr = (vr + (1 << 11)) >> 12; 2945 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2946 2947 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2948 2949 // XXX: these things DON'T need to be done each time 2950 mAudioMixer->setBufferProvider(name, track); 2951 mAudioMixer->enable(name); 2952 2953 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2954 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2955 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2956 mAudioMixer->setParameter( 2957 name, 2958 AudioMixer::TRACK, 2959 AudioMixer::FORMAT, (void *)track->format()); 2960 mAudioMixer->setParameter( 2961 name, 2962 AudioMixer::TRACK, 2963 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2964 mAudioMixer->setParameter( 2965 name, 2966 AudioMixer::RESAMPLE, 2967 AudioMixer::SAMPLE_RATE, 2968 (void *)(cblk->sampleRate)); 2969 mAudioMixer->setParameter( 2970 name, 2971 AudioMixer::TRACK, 2972 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2973 mAudioMixer->setParameter( 2974 name, 2975 AudioMixer::TRACK, 2976 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2977 2978 // reset retry count 2979 track->mRetryCount = kMaxTrackRetries; 2980 2981 // If one track is ready, set the mixer ready if: 2982 // - the mixer was not ready during previous round OR 2983 // - no other track is not ready 2984 if (mMixerStatus != MIXER_TRACKS_READY || 2985 mixerStatus != MIXER_TRACKS_ENABLED) { 2986 mixerStatus = MIXER_TRACKS_READY; 2987 } 2988 } else { 2989 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2990 if (track->isStopped()) { 2991 track->reset(); 2992 } 2993 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2994 // We have consumed all the buffers of this track. 2995 // Remove it from the list of active tracks. 2996 // TODO: use actual buffer filling status instead of latency when available from 2997 // audio HAL 2998 size_t audioHALFrames = 2999 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3000 size_t framesWritten = 3001 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3002 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3003 tracksToRemove->add(track); 3004 } 3005 } else { 3006 // No buffers for this track. Give it a few chances to 3007 // fill a buffer, then remove it from active list. 3008 if (--(track->mRetryCount) <= 0) { 3009 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3010 tracksToRemove->add(track); 3011 // indicate to client process that the track was disabled because of underrun 3012 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3013 // If one track is not ready, mark the mixer also not ready if: 3014 // - the mixer was ready during previous round OR 3015 // - no other track is ready 3016 } else if (mMixerStatus == MIXER_TRACKS_READY || 3017 mixerStatus != MIXER_TRACKS_READY) { 3018 mixerStatus = MIXER_TRACKS_ENABLED; 3019 } 3020 } 3021 mAudioMixer->disable(name); 3022 } 3023 3024 } // local variable scope to avoid goto warning 3025track_is_ready: ; 3026 3027 } 3028 3029 // FIXME Here is where we would push the new FastMixer state if necessary 3030 3031 // remove all the tracks that need to be... 3032 count = tracksToRemove->size(); 3033 if (CC_UNLIKELY(count)) { 3034 for (size_t i=0 ; i<count ; i++) { 3035 const sp<Track>& track = tracksToRemove->itemAt(i); 3036 mActiveTracks.remove(track); 3037 if (track->mainBuffer() != mMixBuffer) { 3038 chain = getEffectChain_l(track->sessionId()); 3039 if (chain != 0) { 3040 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3041 chain->decActiveTrackCnt(); 3042 } 3043 } 3044 if (track->isTerminated()) { 3045 removeTrack_l(track); 3046 } 3047 } 3048 } 3049 3050 // mix buffer must be cleared if all tracks are connected to an 3051 // effect chain as in this case the mixer will not write to 3052 // mix buffer and track effects will accumulate into it 3053 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3054 // FIXME as a performance optimization, should remember previous zero status 3055 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3056 } 3057 3058 // if any fast tracks, then status is ready 3059 if (fastTracks > 0) { 3060 mixerStatus = MIXER_TRACKS_READY; 3061 } 3062 return mixerStatus; 3063} 3064 3065/* 3066The derived values that are cached: 3067 - mixBufferSize from frame count * frame size 3068 - activeSleepTime from activeSleepTimeUs() 3069 - idleSleepTime from idleSleepTimeUs() 3070 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3071 - maxPeriod from frame count and sample rate (MIXER only) 3072 3073The parameters that affect these derived values are: 3074 - frame count 3075 - frame size 3076 - sample rate 3077 - device type: A2DP or not 3078 - device latency 3079 - format: PCM or not 3080 - active sleep time 3081 - idle sleep time 3082*/ 3083 3084void AudioFlinger::PlaybackThread::cacheParameters_l() 3085{ 3086 mixBufferSize = mNormalFrameCount * mFrameSize; 3087 activeSleepTime = activeSleepTimeUs(); 3088 idleSleepTime = idleSleepTimeUs(); 3089} 3090 3091void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3092{ 3093 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3094 this, streamType, mTracks.size()); 3095 Mutex::Autolock _l(mLock); 3096 3097 size_t size = mTracks.size(); 3098 for (size_t i = 0; i < size; i++) { 3099 sp<Track> t = mTracks[i]; 3100 if (t->streamType() == streamType) { 3101 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3102 t->mCblk->cv.signal(); 3103 } 3104 } 3105} 3106 3107// getTrackName_l() must be called with ThreadBase::mLock held 3108int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3109{ 3110 return mAudioMixer->getTrackName(channelMask); 3111} 3112 3113// deleteTrackName_l() must be called with ThreadBase::mLock held 3114void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3115{ 3116 ALOGV("remove track (%d) and delete from mixer", name); 3117 mAudioMixer->deleteTrackName(name); 3118} 3119 3120// checkForNewParameters_l() must be called with ThreadBase::mLock held 3121bool AudioFlinger::MixerThread::checkForNewParameters_l() 3122{ 3123 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3124 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3125 bool reconfig = false; 3126 3127 while (!mNewParameters.isEmpty()) { 3128 3129 if (mFastMixer != NULL) { 3130 FastMixerStateQueue *sq = mFastMixer->sq(); 3131 FastMixerState *state = sq->begin(); 3132 if (!(state->mCommand & FastMixerState::IDLE)) { 3133 previousCommand = state->mCommand; 3134 state->mCommand = FastMixerState::HOT_IDLE; 3135 sq->end(); 3136 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3137 } else { 3138 sq->end(false /*didModify*/); 3139 } 3140 } 3141 3142 status_t status = NO_ERROR; 3143 String8 keyValuePair = mNewParameters[0]; 3144 AudioParameter param = AudioParameter(keyValuePair); 3145 int value; 3146 3147 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3148 reconfig = true; 3149 } 3150 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3151 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3152 status = BAD_VALUE; 3153 } else { 3154 reconfig = true; 3155 } 3156 } 3157 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3158 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3159 status = BAD_VALUE; 3160 } else { 3161 reconfig = true; 3162 } 3163 } 3164 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3165 // do not accept frame count changes if tracks are open as the track buffer 3166 // size depends on frame count and correct behavior would not be guaranteed 3167 // if frame count is changed after track creation 3168 if (!mTracks.isEmpty()) { 3169 status = INVALID_OPERATION; 3170 } else { 3171 reconfig = true; 3172 } 3173 } 3174 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3175#ifdef ADD_BATTERY_DATA 3176 // when changing the audio output device, call addBatteryData to notify 3177 // the change 3178 if ((int)mDevice != value) { 3179 uint32_t params = 0; 3180 // check whether speaker is on 3181 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3182 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3183 } 3184 3185 int deviceWithoutSpeaker 3186 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3187 // check if any other device (except speaker) is on 3188 if (value & deviceWithoutSpeaker ) { 3189 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3190 } 3191 3192 if (params != 0) { 3193 addBatteryData(params); 3194 } 3195 } 3196#endif 3197 3198 // forward device change to effects that have requested to be 3199 // aware of attached audio device. 3200 mDevice = (uint32_t)value; 3201 for (size_t i = 0; i < mEffectChains.size(); i++) { 3202 mEffectChains[i]->setDevice_l(mDevice); 3203 } 3204 } 3205 3206 if (status == NO_ERROR) { 3207 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3208 keyValuePair.string()); 3209 if (!mStandby && status == INVALID_OPERATION) { 3210 mOutput->stream->common.standby(&mOutput->stream->common); 3211 mStandby = true; 3212 mBytesWritten = 0; 3213 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3214 keyValuePair.string()); 3215 } 3216 if (status == NO_ERROR && reconfig) { 3217 delete mAudioMixer; 3218 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3219 mAudioMixer = NULL; 3220 readOutputParameters(); 3221 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3222 for (size_t i = 0; i < mTracks.size() ; i++) { 3223 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3224 if (name < 0) break; 3225 mTracks[i]->mName = name; 3226 // limit track sample rate to 2 x new output sample rate 3227 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3228 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3229 } 3230 } 3231 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3232 } 3233 } 3234 3235 mNewParameters.removeAt(0); 3236 3237 mParamStatus = status; 3238 mParamCond.signal(); 3239 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3240 // already timed out waiting for the status and will never signal the condition. 3241 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3242 } 3243 3244 if (!(previousCommand & FastMixerState::IDLE)) { 3245 ALOG_ASSERT(mFastMixer != NULL); 3246 FastMixerStateQueue *sq = mFastMixer->sq(); 3247 FastMixerState *state = sq->begin(); 3248 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3249 state->mCommand = previousCommand; 3250 sq->end(); 3251 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3252 } 3253 3254 return reconfig; 3255} 3256 3257status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3258{ 3259 const size_t SIZE = 256; 3260 char buffer[SIZE]; 3261 String8 result; 3262 3263 PlaybackThread::dumpInternals(fd, args); 3264 3265 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3266 result.append(buffer); 3267 write(fd, result.string(), result.size()); 3268 3269 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3270 FastMixerDumpState copy = mFastMixerDumpState; 3271 copy.dump(fd); 3272 3273 return NO_ERROR; 3274} 3275 3276uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3277{ 3278 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3279} 3280 3281uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3282{ 3283 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3284} 3285 3286void AudioFlinger::MixerThread::cacheParameters_l() 3287{ 3288 PlaybackThread::cacheParameters_l(); 3289 3290 // FIXME: Relaxed timing because of a certain device that can't meet latency 3291 // Should be reduced to 2x after the vendor fixes the driver issue 3292 // increase threshold again due to low power audio mode. The way this warning 3293 // threshold is calculated and its usefulness should be reconsidered anyway. 3294 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3295} 3296 3297// ---------------------------------------------------------------------------- 3298AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3299 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3300 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3301 // mLeftVolFloat, mRightVolFloat 3302 // mLeftVolShort, mRightVolShort 3303{ 3304} 3305 3306AudioFlinger::DirectOutputThread::~DirectOutputThread() 3307{ 3308} 3309 3310AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3311 Vector< sp<Track> > *tracksToRemove 3312) 3313{ 3314 sp<Track> trackToRemove; 3315 3316 mixer_state mixerStatus = MIXER_IDLE; 3317 3318 // find out which tracks need to be processed 3319 if (mActiveTracks.size() != 0) { 3320 sp<Track> t = mActiveTracks[0].promote(); 3321 // The track died recently 3322 if (t == 0) return MIXER_IDLE; 3323 3324 Track* const track = t.get(); 3325 audio_track_cblk_t* cblk = track->cblk(); 3326 3327 // The first time a track is added we wait 3328 // for all its buffers to be filled before processing it 3329 if (cblk->framesReady() && track->isReady() && 3330 !track->isPaused() && !track->isTerminated()) 3331 { 3332 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3333 3334 if (track->mFillingUpStatus == Track::FS_FILLED) { 3335 track->mFillingUpStatus = Track::FS_ACTIVE; 3336 mLeftVolFloat = mRightVolFloat = 0; 3337 mLeftVolShort = mRightVolShort = 0; 3338 if (track->mState == TrackBase::RESUMING) { 3339 track->mState = TrackBase::ACTIVE; 3340 rampVolume = true; 3341 } 3342 } else if (cblk->server != 0) { 3343 // If the track is stopped before the first frame was mixed, 3344 // do not apply ramp 3345 rampVolume = true; 3346 } 3347 // compute volume for this track 3348 float left, right; 3349 if (track->isMuted() || mMasterMute || track->isPausing() || 3350 mStreamTypes[track->streamType()].mute) { 3351 left = right = 0; 3352 if (track->isPausing()) { 3353 track->setPaused(); 3354 } 3355 } else { 3356 float typeVolume = mStreamTypes[track->streamType()].volume; 3357 float v = mMasterVolume * typeVolume; 3358 uint32_t vlr = cblk->getVolumeLR(); 3359 float v_clamped = v * (vlr & 0xFFFF); 3360 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3361 left = v_clamped/MAX_GAIN; 3362 v_clamped = v * (vlr >> 16); 3363 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3364 right = v_clamped/MAX_GAIN; 3365 } 3366 3367 if (left != mLeftVolFloat || right != mRightVolFloat) { 3368 mLeftVolFloat = left; 3369 mRightVolFloat = right; 3370 3371 // If audio HAL implements volume control, 3372 // force software volume to nominal value 3373 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3374 left = 1.0f; 3375 right = 1.0f; 3376 } 3377 3378 // Convert volumes from float to 8.24 3379 uint32_t vl = (uint32_t)(left * (1 << 24)); 3380 uint32_t vr = (uint32_t)(right * (1 << 24)); 3381 3382 // Delegate volume control to effect in track effect chain if needed 3383 // only one effect chain can be present on DirectOutputThread, so if 3384 // there is one, the track is connected to it 3385 if (!mEffectChains.isEmpty()) { 3386 // Do not ramp volume if volume is controlled by effect 3387 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3388 rampVolume = false; 3389 } 3390 } 3391 3392 // Convert volumes from 8.24 to 4.12 format 3393 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3394 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3395 leftVol = (uint16_t)v_clamped; 3396 v_clamped = (vr + (1 << 11)) >> 12; 3397 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3398 rightVol = (uint16_t)v_clamped; 3399 } else { 3400 leftVol = mLeftVolShort; 3401 rightVol = mRightVolShort; 3402 rampVolume = false; 3403 } 3404 3405 // reset retry count 3406 track->mRetryCount = kMaxTrackRetriesDirect; 3407 mActiveTrack = t; 3408 mixerStatus = MIXER_TRACKS_READY; 3409 } else { 3410 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3411 if (track->isStopped()) { 3412 track->reset(); 3413 } 3414 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3415 // We have consumed all the buffers of this track. 3416 // Remove it from the list of active tracks. 3417 // TODO: implement behavior for compressed audio 3418 size_t audioHALFrames = 3419 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3420 size_t framesWritten = 3421 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3422 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3423 trackToRemove = track; 3424 } 3425 } else { 3426 // No buffers for this track. Give it a few chances to 3427 // fill a buffer, then remove it from active list. 3428 if (--(track->mRetryCount) <= 0) { 3429 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3430 trackToRemove = track; 3431 } else { 3432 mixerStatus = MIXER_TRACKS_ENABLED; 3433 } 3434 } 3435 } 3436 } 3437 3438 // FIXME merge this with similar code for removing multiple tracks 3439 // remove all the tracks that need to be... 3440 if (CC_UNLIKELY(trackToRemove != 0)) { 3441 tracksToRemove->add(trackToRemove); 3442 mActiveTracks.remove(trackToRemove); 3443 if (!mEffectChains.isEmpty()) { 3444 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3445 trackToRemove->sessionId()); 3446 mEffectChains[0]->decActiveTrackCnt(); 3447 } 3448 if (trackToRemove->isTerminated()) { 3449 removeTrack_l(trackToRemove); 3450 } 3451 } 3452 3453 return mixerStatus; 3454} 3455 3456void AudioFlinger::DirectOutputThread::threadLoop_mix() 3457{ 3458 AudioBufferProvider::Buffer buffer; 3459 size_t frameCount = mFrameCount; 3460 int8_t *curBuf = (int8_t *)mMixBuffer; 3461 // output audio to hardware 3462 while (frameCount) { 3463 buffer.frameCount = frameCount; 3464 mActiveTrack->getNextBuffer(&buffer); 3465 if (CC_UNLIKELY(buffer.raw == NULL)) { 3466 memset(curBuf, 0, frameCount * mFrameSize); 3467 break; 3468 } 3469 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3470 frameCount -= buffer.frameCount; 3471 curBuf += buffer.frameCount * mFrameSize; 3472 mActiveTrack->releaseBuffer(&buffer); 3473 } 3474 sleepTime = 0; 3475 standbyTime = systemTime() + standbyDelay; 3476 mActiveTrack.clear(); 3477 3478 // apply volume 3479 3480 // Do not apply volume on compressed audio 3481 if (!audio_is_linear_pcm(mFormat)) { 3482 return; 3483 } 3484 3485 // convert to signed 16 bit before volume calculation 3486 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3487 size_t count = mFrameCount * mChannelCount; 3488 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3489 int16_t *dst = mMixBuffer + count-1; 3490 while (count--) { 3491 *dst-- = (int16_t)(*src--^0x80) << 8; 3492 } 3493 } 3494 3495 frameCount = mFrameCount; 3496 int16_t *out = mMixBuffer; 3497 if (rampVolume) { 3498 if (mChannelCount == 1) { 3499 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3500 int32_t vlInc = d / (int32_t)frameCount; 3501 int32_t vl = ((int32_t)mLeftVolShort << 16); 3502 do { 3503 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3504 out++; 3505 vl += vlInc; 3506 } while (--frameCount); 3507 3508 } else { 3509 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3510 int32_t vlInc = d / (int32_t)frameCount; 3511 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3512 int32_t vrInc = d / (int32_t)frameCount; 3513 int32_t vl = ((int32_t)mLeftVolShort << 16); 3514 int32_t vr = ((int32_t)mRightVolShort << 16); 3515 do { 3516 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3517 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3518 out += 2; 3519 vl += vlInc; 3520 vr += vrInc; 3521 } while (--frameCount); 3522 } 3523 } else { 3524 if (mChannelCount == 1) { 3525 do { 3526 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3527 out++; 3528 } while (--frameCount); 3529 } else { 3530 do { 3531 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3532 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3533 out += 2; 3534 } while (--frameCount); 3535 } 3536 } 3537 3538 // convert back to unsigned 8 bit after volume calculation 3539 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3540 size_t count = mFrameCount * mChannelCount; 3541 int16_t *src = mMixBuffer; 3542 uint8_t *dst = (uint8_t *)mMixBuffer; 3543 while (count--) { 3544 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3545 } 3546 } 3547 3548 mLeftVolShort = leftVol; 3549 mRightVolShort = rightVol; 3550} 3551 3552void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3553{ 3554 if (sleepTime == 0) { 3555 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3556 sleepTime = activeSleepTime; 3557 } else { 3558 sleepTime = idleSleepTime; 3559 } 3560 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3561 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3562 sleepTime = 0; 3563 } 3564} 3565 3566// getTrackName_l() must be called with ThreadBase::mLock held 3567int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3568{ 3569 return 0; 3570} 3571 3572// deleteTrackName_l() must be called with ThreadBase::mLock held 3573void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3574{ 3575} 3576 3577// checkForNewParameters_l() must be called with ThreadBase::mLock held 3578bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3579{ 3580 bool reconfig = false; 3581 3582 while (!mNewParameters.isEmpty()) { 3583 status_t status = NO_ERROR; 3584 String8 keyValuePair = mNewParameters[0]; 3585 AudioParameter param = AudioParameter(keyValuePair); 3586 int value; 3587 3588 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3589 // do not accept frame count changes if tracks are open as the track buffer 3590 // size depends on frame count and correct behavior would not be garantied 3591 // if frame count is changed after track creation 3592 if (!mTracks.isEmpty()) { 3593 status = INVALID_OPERATION; 3594 } else { 3595 reconfig = true; 3596 } 3597 } 3598 if (status == NO_ERROR) { 3599 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3600 keyValuePair.string()); 3601 if (!mStandby && status == INVALID_OPERATION) { 3602 mOutput->stream->common.standby(&mOutput->stream->common); 3603 mStandby = true; 3604 mBytesWritten = 0; 3605 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3606 keyValuePair.string()); 3607 } 3608 if (status == NO_ERROR && reconfig) { 3609 readOutputParameters(); 3610 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3611 } 3612 } 3613 3614 mNewParameters.removeAt(0); 3615 3616 mParamStatus = status; 3617 mParamCond.signal(); 3618 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3619 // already timed out waiting for the status and will never signal the condition. 3620 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3621 } 3622 return reconfig; 3623} 3624 3625uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3626{ 3627 uint32_t time; 3628 if (audio_is_linear_pcm(mFormat)) { 3629 time = PlaybackThread::activeSleepTimeUs(); 3630 } else { 3631 time = 10000; 3632 } 3633 return time; 3634} 3635 3636uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3637{ 3638 uint32_t time; 3639 if (audio_is_linear_pcm(mFormat)) { 3640 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3641 } else { 3642 time = 10000; 3643 } 3644 return time; 3645} 3646 3647uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3648{ 3649 uint32_t time; 3650 if (audio_is_linear_pcm(mFormat)) { 3651 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3652 } else { 3653 time = 10000; 3654 } 3655 return time; 3656} 3657 3658void AudioFlinger::DirectOutputThread::cacheParameters_l() 3659{ 3660 PlaybackThread::cacheParameters_l(); 3661 3662 // use shorter standby delay as on normal output to release 3663 // hardware resources as soon as possible 3664 standbyDelay = microseconds(activeSleepTime*2); 3665} 3666 3667// ---------------------------------------------------------------------------- 3668 3669AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3670 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3671 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3672 mWaitTimeMs(UINT_MAX) 3673{ 3674 addOutputTrack(mainThread); 3675} 3676 3677AudioFlinger::DuplicatingThread::~DuplicatingThread() 3678{ 3679 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3680 mOutputTracks[i]->destroy(); 3681 } 3682} 3683 3684void AudioFlinger::DuplicatingThread::threadLoop_mix() 3685{ 3686 // mix buffers... 3687 if (outputsReady(outputTracks)) { 3688 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3689 } else { 3690 memset(mMixBuffer, 0, mixBufferSize); 3691 } 3692 sleepTime = 0; 3693 writeFrames = mNormalFrameCount; 3694} 3695 3696void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3697{ 3698 if (sleepTime == 0) { 3699 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3700 sleepTime = activeSleepTime; 3701 } else { 3702 sleepTime = idleSleepTime; 3703 } 3704 } else if (mBytesWritten != 0) { 3705 // flush remaining overflow buffers in output tracks 3706 for (size_t i = 0; i < outputTracks.size(); i++) { 3707 if (outputTracks[i]->isActive()) { 3708 sleepTime = 0; 3709 writeFrames = 0; 3710 memset(mMixBuffer, 0, mixBufferSize); 3711 break; 3712 } 3713 } 3714 } 3715} 3716 3717void AudioFlinger::DuplicatingThread::threadLoop_write() 3718{ 3719 standbyTime = systemTime() + standbyDelay; 3720 for (size_t i = 0; i < outputTracks.size(); i++) { 3721 outputTracks[i]->write(mMixBuffer, writeFrames); 3722 } 3723 mBytesWritten += mixBufferSize; 3724} 3725 3726void AudioFlinger::DuplicatingThread::threadLoop_standby() 3727{ 3728 // DuplicatingThread implements standby by stopping all tracks 3729 for (size_t i = 0; i < outputTracks.size(); i++) { 3730 outputTracks[i]->stop(); 3731 } 3732} 3733 3734void AudioFlinger::DuplicatingThread::saveOutputTracks() 3735{ 3736 outputTracks = mOutputTracks; 3737} 3738 3739void AudioFlinger::DuplicatingThread::clearOutputTracks() 3740{ 3741 outputTracks.clear(); 3742} 3743 3744void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3745{ 3746 Mutex::Autolock _l(mLock); 3747 // FIXME explain this formula 3748 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3749 OutputTrack *outputTrack = new OutputTrack(thread, 3750 this, 3751 mSampleRate, 3752 mFormat, 3753 mChannelMask, 3754 frameCount); 3755 if (outputTrack->cblk() != NULL) { 3756 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3757 mOutputTracks.add(outputTrack); 3758 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3759 updateWaitTime_l(); 3760 } 3761} 3762 3763void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3764{ 3765 Mutex::Autolock _l(mLock); 3766 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3767 if (mOutputTracks[i]->thread() == thread) { 3768 mOutputTracks[i]->destroy(); 3769 mOutputTracks.removeAt(i); 3770 updateWaitTime_l(); 3771 return; 3772 } 3773 } 3774 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3775} 3776 3777// caller must hold mLock 3778void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3779{ 3780 mWaitTimeMs = UINT_MAX; 3781 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3782 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3783 if (strong != 0) { 3784 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3785 if (waitTimeMs < mWaitTimeMs) { 3786 mWaitTimeMs = waitTimeMs; 3787 } 3788 } 3789 } 3790} 3791 3792 3793bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3794{ 3795 for (size_t i = 0; i < outputTracks.size(); i++) { 3796 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3797 if (thread == 0) { 3798 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3799 return false; 3800 } 3801 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3802 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3803 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3804 return false; 3805 } 3806 } 3807 return true; 3808} 3809 3810uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3811{ 3812 return (mWaitTimeMs * 1000) / 2; 3813} 3814 3815void AudioFlinger::DuplicatingThread::cacheParameters_l() 3816{ 3817 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3818 updateWaitTime_l(); 3819 3820 MixerThread::cacheParameters_l(); 3821} 3822 3823// ---------------------------------------------------------------------------- 3824 3825// TrackBase constructor must be called with AudioFlinger::mLock held 3826AudioFlinger::ThreadBase::TrackBase::TrackBase( 3827 ThreadBase *thread, 3828 const sp<Client>& client, 3829 uint32_t sampleRate, 3830 audio_format_t format, 3831 uint32_t channelMask, 3832 int frameCount, 3833 const sp<IMemory>& sharedBuffer, 3834 int sessionId) 3835 : RefBase(), 3836 mThread(thread), 3837 mClient(client), 3838 mCblk(NULL), 3839 // mBuffer 3840 // mBufferEnd 3841 mFrameCount(0), 3842 mState(IDLE), 3843 mSampleRate(sampleRate), 3844 mFormat(format), 3845 mStepServerFailed(false), 3846 mSessionId(sessionId) 3847 // mChannelCount 3848 // mChannelMask 3849{ 3850 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3851 3852 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3853 size_t size = sizeof(audio_track_cblk_t); 3854 uint8_t channelCount = popcount(channelMask); 3855 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3856 if (sharedBuffer == 0) { 3857 size += bufferSize; 3858 } 3859 3860 if (client != NULL) { 3861 mCblkMemory = client->heap()->allocate(size); 3862 if (mCblkMemory != 0) { 3863 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3864 if (mCblk != NULL) { // construct the shared structure in-place. 3865 new(mCblk) audio_track_cblk_t(); 3866 // clear all buffers 3867 mCblk->frameCount = frameCount; 3868 mCblk->sampleRate = sampleRate; 3869// uncomment the following lines to quickly test 32-bit wraparound 3870// mCblk->user = 0xffff0000; 3871// mCblk->server = 0xffff0000; 3872// mCblk->userBase = 0xffff0000; 3873// mCblk->serverBase = 0xffff0000; 3874 mChannelCount = channelCount; 3875 mChannelMask = channelMask; 3876 if (sharedBuffer == 0) { 3877 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3878 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3879 // Force underrun condition to avoid false underrun callback until first data is 3880 // written to buffer (other flags are cleared) 3881 mCblk->flags = CBLK_UNDERRUN_ON; 3882 } else { 3883 mBuffer = sharedBuffer->pointer(); 3884 } 3885 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3886 } 3887 } else { 3888 ALOGE("not enough memory for AudioTrack size=%u", size); 3889 client->heap()->dump("AudioTrack"); 3890 return; 3891 } 3892 } else { 3893 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3894 // construct the shared structure in-place. 3895 new(mCblk) audio_track_cblk_t(); 3896 // clear all buffers 3897 mCblk->frameCount = frameCount; 3898 mCblk->sampleRate = sampleRate; 3899// uncomment the following lines to quickly test 32-bit wraparound 3900// mCblk->user = 0xffff0000; 3901// mCblk->server = 0xffff0000; 3902// mCblk->userBase = 0xffff0000; 3903// mCblk->serverBase = 0xffff0000; 3904 mChannelCount = channelCount; 3905 mChannelMask = channelMask; 3906 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3907 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3908 // Force underrun condition to avoid false underrun callback until first data is 3909 // written to buffer (other flags are cleared) 3910 mCblk->flags = CBLK_UNDERRUN_ON; 3911 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3912 } 3913} 3914 3915AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3916{ 3917 if (mCblk != NULL) { 3918 if (mClient == 0) { 3919 delete mCblk; 3920 } else { 3921 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3922 } 3923 } 3924 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3925 if (mClient != 0) { 3926 // Client destructor must run with AudioFlinger mutex locked 3927 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3928 // If the client's reference count drops to zero, the associated destructor 3929 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3930 // relying on the automatic clear() at end of scope. 3931 mClient.clear(); 3932 } 3933} 3934 3935// AudioBufferProvider interface 3936// getNextBuffer() = 0; 3937// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3938void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3939{ 3940 buffer->raw = NULL; 3941 mFrameCount = buffer->frameCount; 3942 (void) step(); // ignore return value of step() 3943 buffer->frameCount = 0; 3944} 3945 3946bool AudioFlinger::ThreadBase::TrackBase::step() { 3947 bool result; 3948 audio_track_cblk_t* cblk = this->cblk(); 3949 3950 result = cblk->stepServer(mFrameCount); 3951 if (!result) { 3952 ALOGV("stepServer failed acquiring cblk mutex"); 3953 mStepServerFailed = true; 3954 } 3955 return result; 3956} 3957 3958void AudioFlinger::ThreadBase::TrackBase::reset() { 3959 audio_track_cblk_t* cblk = this->cblk(); 3960 3961 cblk->user = 0; 3962 cblk->server = 0; 3963 cblk->userBase = 0; 3964 cblk->serverBase = 0; 3965 mStepServerFailed = false; 3966 ALOGV("TrackBase::reset"); 3967} 3968 3969int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3970 return (int)mCblk->sampleRate; 3971} 3972 3973void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3974 audio_track_cblk_t* cblk = this->cblk(); 3975 size_t frameSize = cblk->frameSize; 3976 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3977 int8_t *bufferEnd = bufferStart + frames * frameSize; 3978 3979 // Check validity of returned pointer in case the track control block would have been corrupted. 3980 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 3981 "TrackBase::getBuffer buffer out of range:\n" 3982 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 3983 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 3984 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3985 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 3986 3987 return bufferStart; 3988} 3989 3990status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3991{ 3992 mSyncEvents.add(event); 3993 return NO_ERROR; 3994} 3995 3996// ---------------------------------------------------------------------------- 3997 3998// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3999AudioFlinger::PlaybackThread::Track::Track( 4000 PlaybackThread *thread, 4001 const sp<Client>& client, 4002 audio_stream_type_t streamType, 4003 uint32_t sampleRate, 4004 audio_format_t format, 4005 uint32_t channelMask, 4006 int frameCount, 4007 const sp<IMemory>& sharedBuffer, 4008 int sessionId, 4009 IAudioFlinger::track_flags_t flags) 4010 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4011 mMute(false), 4012 mFillingUpStatus(FS_INVALID), 4013 // mRetryCount initialized later when needed 4014 mSharedBuffer(sharedBuffer), 4015 mStreamType(streamType), 4016 mName(-1), // see note below 4017 mMainBuffer(thread->mixBuffer()), 4018 mAuxBuffer(NULL), 4019 mAuxEffectId(0), mHasVolumeController(false), 4020 mPresentationCompleteFrames(0), 4021 mFlags(flags), 4022 mFastIndex(-1), 4023 mCachedVolume(1.0) 4024{ 4025 if (mCblk != NULL) { 4026 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4027 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4028 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4029 if (flags & IAudioFlinger::TRACK_FAST) { 4030 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4031 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4032 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4033 ALOG_ASSERT(0 < i && i < FastMixerState::kMaxFastTracks); 4034 mFastIndex = i; 4035 thread->mFastTrackAvailMask &= ~(1 << i); 4036 // Although we've allocated an index, we can't mutate or push a new fast track state 4037 // here, because that data structure can only be changed within the normal mixer 4038 // threadLoop(). So instead, make a note to mutate and push later. 4039 thread->mFastTrackNewArray[i] = this; 4040 thread->mFastTrackNewMask |= 1 << i; 4041 } 4042 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4043 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4044 if (mName < 0) { 4045 ALOGE("no more track names available"); 4046 } 4047 } 4048 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4049} 4050 4051AudioFlinger::PlaybackThread::Track::~Track() 4052{ 4053 ALOGV("PlaybackThread::Track destructor"); 4054 sp<ThreadBase> thread = mThread.promote(); 4055 if (thread != 0) { 4056 Mutex::Autolock _l(thread->mLock); 4057 mState = TERMINATED; 4058 } 4059} 4060 4061void AudioFlinger::PlaybackThread::Track::destroy() 4062{ 4063 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4064 // by removing it from mTracks vector, so there is a risk that this Tracks's 4065 // destructor is called. As the destructor needs to lock mLock, 4066 // we must acquire a strong reference on this Track before locking mLock 4067 // here so that the destructor is called only when exiting this function. 4068 // On the other hand, as long as Track::destroy() is only called by 4069 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4070 // this Track with its member mTrack. 4071 sp<Track> keep(this); 4072 { // scope for mLock 4073 sp<ThreadBase> thread = mThread.promote(); 4074 if (thread != 0) { 4075 if (!isOutputTrack()) { 4076 if (mState == ACTIVE || mState == RESUMING) { 4077 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4078 4079#ifdef ADD_BATTERY_DATA 4080 // to track the speaker usage 4081 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4082#endif 4083 } 4084 AudioSystem::releaseOutput(thread->id()); 4085 } 4086 Mutex::Autolock _l(thread->mLock); 4087 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4088 playbackThread->destroyTrack_l(this); 4089 } 4090 } 4091} 4092 4093void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4094{ 4095 uint32_t vlr = mCblk->getVolumeLR(); 4096 if (isFastTrack()) { 4097 strcpy(buffer, " fast"); 4098 } else { 4099 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4100 } 4101 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", 4102 (mClient == 0) ? getpid_cached : mClient->pid(), 4103 mStreamType, 4104 mFormat, 4105 mChannelMask, 4106 mSessionId, 4107 mFrameCount, 4108 mState, 4109 mMute, 4110 mFillingUpStatus, 4111 mCblk->sampleRate, 4112 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4113 20.0 * log10((vlr >> 16) / 4096.0), 4114 mCblk->server, 4115 mCblk->user, 4116 (int)mMainBuffer, 4117 (int)mAuxBuffer); 4118} 4119 4120// AudioBufferProvider interface 4121status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4122 AudioBufferProvider::Buffer* buffer, int64_t pts) 4123{ 4124 audio_track_cblk_t* cblk = this->cblk(); 4125 uint32_t framesReady; 4126 uint32_t framesReq = buffer->frameCount; 4127 4128 // Check if last stepServer failed, try to step now 4129 if (mStepServerFailed) { 4130 if (!step()) goto getNextBuffer_exit; 4131 ALOGV("stepServer recovered"); 4132 mStepServerFailed = false; 4133 } 4134 4135 framesReady = cblk->framesReady(); 4136 4137 if (CC_LIKELY(framesReady)) { 4138 uint32_t s = cblk->server; 4139 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4140 4141 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4142 if (framesReq > framesReady) { 4143 framesReq = framesReady; 4144 } 4145 if (framesReq > bufferEnd - s) { 4146 framesReq = bufferEnd - s; 4147 } 4148 4149 buffer->raw = getBuffer(s, framesReq); 4150 if (buffer->raw == NULL) goto getNextBuffer_exit; 4151 4152 buffer->frameCount = framesReq; 4153 return NO_ERROR; 4154 } 4155 4156getNextBuffer_exit: 4157 buffer->raw = NULL; 4158 buffer->frameCount = 0; 4159 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4160 return NOT_ENOUGH_DATA; 4161} 4162 4163uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4164 return mCblk->framesReady(); 4165} 4166 4167bool AudioFlinger::PlaybackThread::Track::isReady() const { 4168 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4169 4170 if (framesReady() >= mCblk->frameCount || 4171 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4172 mFillingUpStatus = FS_FILLED; 4173 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4174 return true; 4175 } 4176 return false; 4177} 4178 4179status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4180 int triggerSession) 4181{ 4182 status_t status = NO_ERROR; 4183 ALOGV("start(%d), calling pid %d session %d", 4184 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4185 4186 sp<ThreadBase> thread = mThread.promote(); 4187 if (thread != 0) { 4188 Mutex::Autolock _l(thread->mLock); 4189 track_state state = mState; 4190 // here the track could be either new, or restarted 4191 // in both cases "unstop" the track 4192 if (mState == PAUSED) { 4193 mState = TrackBase::RESUMING; 4194 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4195 } else { 4196 mState = TrackBase::ACTIVE; 4197 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4198 } 4199 4200 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4201 thread->mLock.unlock(); 4202 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4203 thread->mLock.lock(); 4204 4205#ifdef ADD_BATTERY_DATA 4206 // to track the speaker usage 4207 if (status == NO_ERROR) { 4208 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4209 } 4210#endif 4211 } 4212 if (status == NO_ERROR) { 4213 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4214 playbackThread->addTrack_l(this); 4215 } else { 4216 mState = state; 4217 } 4218 } else { 4219 status = BAD_VALUE; 4220 } 4221 return status; 4222} 4223 4224void AudioFlinger::PlaybackThread::Track::stop() 4225{ 4226 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4227 sp<ThreadBase> thread = mThread.promote(); 4228 if (thread != 0) { 4229 Mutex::Autolock _l(thread->mLock); 4230 track_state state = mState; 4231 if (mState > STOPPED) { 4232 mState = STOPPED; 4233 // If the track is not active (PAUSED and buffers full), flush buffers 4234 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4235 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4236 reset(); 4237 } 4238 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 4239 } 4240 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4241 thread->mLock.unlock(); 4242 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4243 thread->mLock.lock(); 4244 4245#ifdef ADD_BATTERY_DATA 4246 // to track the speaker usage 4247 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4248#endif 4249 } 4250 } 4251} 4252 4253void AudioFlinger::PlaybackThread::Track::pause() 4254{ 4255 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4256 sp<ThreadBase> thread = mThread.promote(); 4257 if (thread != 0) { 4258 Mutex::Autolock _l(thread->mLock); 4259 if (mState == ACTIVE || mState == RESUMING) { 4260 mState = PAUSING; 4261 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4262 if (!isOutputTrack()) { 4263 thread->mLock.unlock(); 4264 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4265 thread->mLock.lock(); 4266 4267#ifdef ADD_BATTERY_DATA 4268 // to track the speaker usage 4269 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4270#endif 4271 } 4272 } 4273 } 4274} 4275 4276void AudioFlinger::PlaybackThread::Track::flush() 4277{ 4278 ALOGV("flush(%d)", mName); 4279 sp<ThreadBase> thread = mThread.promote(); 4280 if (thread != 0) { 4281 Mutex::Autolock _l(thread->mLock); 4282 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 4283 return; 4284 } 4285 // No point remaining in PAUSED state after a flush => go to 4286 // STOPPED state 4287 mState = STOPPED; 4288 4289 // do not reset the track if it is still in the process of being stopped or paused. 4290 // this will be done by prepareTracks_l() when the track is stopped. 4291 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4292 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4293 reset(); 4294 } 4295 } 4296} 4297 4298void AudioFlinger::PlaybackThread::Track::reset() 4299{ 4300 // Do not reset twice to avoid discarding data written just after a flush and before 4301 // the audioflinger thread detects the track is stopped. 4302 if (!mResetDone) { 4303 TrackBase::reset(); 4304 // Force underrun condition to avoid false underrun callback until first data is 4305 // written to buffer 4306 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4307 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4308 mFillingUpStatus = FS_FILLING; 4309 mResetDone = true; 4310 mPresentationCompleteFrames = 0; 4311 } 4312} 4313 4314void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4315{ 4316 mMute = muted; 4317} 4318 4319status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4320{ 4321 status_t status = DEAD_OBJECT; 4322 sp<ThreadBase> thread = mThread.promote(); 4323 if (thread != 0) { 4324 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4325 status = playbackThread->attachAuxEffect(this, EffectId); 4326 } 4327 return status; 4328} 4329 4330void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4331{ 4332 mAuxEffectId = EffectId; 4333 mAuxBuffer = buffer; 4334} 4335 4336bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4337 size_t audioHalFrames) 4338{ 4339 // a track is considered presented when the total number of frames written to audio HAL 4340 // corresponds to the number of frames written when presentationComplete() is called for the 4341 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4342 if (mPresentationCompleteFrames == 0) { 4343 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4344 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4345 mPresentationCompleteFrames, audioHalFrames); 4346 } 4347 if (framesWritten >= mPresentationCompleteFrames) { 4348 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4349 mSessionId, framesWritten); 4350 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4351 mPresentationCompleteFrames = 0; 4352 return true; 4353 } 4354 return false; 4355} 4356 4357void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4358{ 4359 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4360 if (mSyncEvents[i]->type() == type) { 4361 mSyncEvents[i]->trigger(); 4362 mSyncEvents.removeAt(i); 4363 i--; 4364 } 4365 } 4366} 4367 4368// implement VolumeBufferProvider interface 4369 4370uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4371{ 4372 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4373 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4374 uint32_t vlr = mCblk->getVolumeLR(); 4375 uint32_t vl = vlr & 0xFFFF; 4376 uint32_t vr = vlr >> 16; 4377 // track volumes come from shared memory, so can't be trusted and must be clamped 4378 if (vl > MAX_GAIN_INT) { 4379 vl = MAX_GAIN_INT; 4380 } 4381 if (vr > MAX_GAIN_INT) { 4382 vr = MAX_GAIN_INT; 4383 } 4384 // now apply the cached master volume and stream type volume; 4385 // this is trusted but lacks any synchronization or barrier so may be stale 4386 float v = mCachedVolume; 4387 vl *= v; 4388 vr *= v; 4389 // re-combine into U4.16 4390 vlr = (vr << 16) | (vl & 0xFFFF); 4391 // FIXME look at mute, pause, and stop flags 4392 return vlr; 4393} 4394 4395// timed audio tracks 4396 4397sp<AudioFlinger::PlaybackThread::TimedTrack> 4398AudioFlinger::PlaybackThread::TimedTrack::create( 4399 PlaybackThread *thread, 4400 const sp<Client>& client, 4401 audio_stream_type_t streamType, 4402 uint32_t sampleRate, 4403 audio_format_t format, 4404 uint32_t channelMask, 4405 int frameCount, 4406 const sp<IMemory>& sharedBuffer, 4407 int sessionId) { 4408 if (!client->reserveTimedTrack()) 4409 return NULL; 4410 4411 return new TimedTrack( 4412 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4413 sharedBuffer, sessionId); 4414} 4415 4416AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4417 PlaybackThread *thread, 4418 const sp<Client>& client, 4419 audio_stream_type_t streamType, 4420 uint32_t sampleRate, 4421 audio_format_t format, 4422 uint32_t channelMask, 4423 int frameCount, 4424 const sp<IMemory>& sharedBuffer, 4425 int sessionId) 4426 : Track(thread, client, streamType, sampleRate, format, channelMask, 4427 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4428 mQueueHeadInFlight(false), 4429 mTrimQueueHeadOnRelease(false), 4430 mFramesPendingInQueue(0), 4431 mTimedSilenceBuffer(NULL), 4432 mTimedSilenceBufferSize(0), 4433 mTimedAudioOutputOnTime(false), 4434 mMediaTimeTransformValid(false) 4435{ 4436 LocalClock lc; 4437 mLocalTimeFreq = lc.getLocalFreq(); 4438 4439 mLocalTimeToSampleTransform.a_zero = 0; 4440 mLocalTimeToSampleTransform.b_zero = 0; 4441 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4442 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4443 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4444 &mLocalTimeToSampleTransform.a_to_b_denom); 4445 4446 mMediaTimeToSampleTransform.a_zero = 0; 4447 mMediaTimeToSampleTransform.b_zero = 0; 4448 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4449 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4450 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4451 &mMediaTimeToSampleTransform.a_to_b_denom); 4452} 4453 4454AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4455 mClient->releaseTimedTrack(); 4456 delete [] mTimedSilenceBuffer; 4457} 4458 4459status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4460 size_t size, sp<IMemory>* buffer) { 4461 4462 Mutex::Autolock _l(mTimedBufferQueueLock); 4463 4464 trimTimedBufferQueue_l(); 4465 4466 // lazily initialize the shared memory heap for timed buffers 4467 if (mTimedMemoryDealer == NULL) { 4468 const int kTimedBufferHeapSize = 512 << 10; 4469 4470 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4471 "AudioFlingerTimed"); 4472 if (mTimedMemoryDealer == NULL) 4473 return NO_MEMORY; 4474 } 4475 4476 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4477 if (newBuffer == NULL) { 4478 newBuffer = mTimedMemoryDealer->allocate(size); 4479 if (newBuffer == NULL) 4480 return NO_MEMORY; 4481 } 4482 4483 *buffer = newBuffer; 4484 return NO_ERROR; 4485} 4486 4487// caller must hold mTimedBufferQueueLock 4488void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4489 int64_t mediaTimeNow; 4490 { 4491 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4492 if (!mMediaTimeTransformValid) 4493 return; 4494 4495 int64_t targetTimeNow; 4496 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4497 ? mCCHelper.getCommonTime(&targetTimeNow) 4498 : mCCHelper.getLocalTime(&targetTimeNow); 4499 4500 if (OK != res) 4501 return; 4502 4503 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4504 &mediaTimeNow)) { 4505 return; 4506 } 4507 } 4508 4509 size_t trimEnd; 4510 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4511 int64_t bufEnd; 4512 4513 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4514 // We have a next buffer. Just use its PTS as the PTS of the frame 4515 // following the last frame in this buffer. If the stream is sparse 4516 // (ie, there are deliberate gaps left in the stream which should be 4517 // filled with silence by the TimedAudioTrack), then this can result 4518 // in one extra buffer being left un-trimmed when it could have 4519 // been. In general, this is not typical, and we would rather 4520 // optimized away the TS calculation below for the more common case 4521 // where PTSes are contiguous. 4522 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4523 } else { 4524 // We have no next buffer. Compute the PTS of the frame following 4525 // the last frame in this buffer by computing the duration of of 4526 // this frame in media time units and adding it to the PTS of the 4527 // buffer. 4528 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4529 / mCblk->frameSize; 4530 4531 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4532 &bufEnd)) { 4533 ALOGE("Failed to convert frame count of %lld to media time" 4534 " duration" " (scale factor %d/%u) in %s", 4535 frameCount, 4536 mMediaTimeToSampleTransform.a_to_b_numer, 4537 mMediaTimeToSampleTransform.a_to_b_denom, 4538 __PRETTY_FUNCTION__); 4539 break; 4540 } 4541 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4542 } 4543 4544 if (bufEnd > mediaTimeNow) 4545 break; 4546 4547 // Is the buffer we want to use in the middle of a mix operation right 4548 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4549 // from the mixer which should be coming back shortly. 4550 if (!trimEnd && mQueueHeadInFlight) { 4551 mTrimQueueHeadOnRelease = true; 4552 } 4553 } 4554 4555 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4556 if (trimStart < trimEnd) { 4557 // Update the bookkeeping for framesReady() 4558 for (size_t i = trimStart; i < trimEnd; ++i) { 4559 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4560 } 4561 4562 // Now actually remove the buffers from the queue. 4563 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4564 } 4565} 4566 4567void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4568 const char* logTag) { 4569 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4570 "%s called (reason \"%s\"), but timed buffer queue has no" 4571 " elements to trim.", __FUNCTION__, logTag); 4572 4573 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4574 mTimedBufferQueue.removeAt(0); 4575} 4576 4577void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4578 const TimedBuffer& buf, 4579 const char* logTag) { 4580 uint32_t bufBytes = buf.buffer()->size(); 4581 uint32_t consumedAlready = buf.position(); 4582 4583 ALOG_ASSERT(consumedAlready <= bufBytes, 4584 "Bad bookkeeping while updating frames pending. Timed buffer is" 4585 " only %u bytes long, but claims to have consumed %u" 4586 " bytes. (update reason: \"%s\")", 4587 bufBytes, consumedAlready, logTag); 4588 4589 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4590 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4591 "Bad bookkeeping while updating frames pending. Should have at" 4592 " least %u queued frames, but we think we have only %u. (update" 4593 " reason: \"%s\")", 4594 bufFrames, mFramesPendingInQueue, logTag); 4595 4596 mFramesPendingInQueue -= bufFrames; 4597} 4598 4599status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4600 const sp<IMemory>& buffer, int64_t pts) { 4601 4602 { 4603 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4604 if (!mMediaTimeTransformValid) 4605 return INVALID_OPERATION; 4606 } 4607 4608 Mutex::Autolock _l(mTimedBufferQueueLock); 4609 4610 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4611 mFramesPendingInQueue += bufFrames; 4612 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4613 4614 return NO_ERROR; 4615} 4616 4617status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4618 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4619 4620 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4621 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4622 target); 4623 4624 if (!(target == TimedAudioTrack::LOCAL_TIME || 4625 target == TimedAudioTrack::COMMON_TIME)) { 4626 return BAD_VALUE; 4627 } 4628 4629 Mutex::Autolock lock(mMediaTimeTransformLock); 4630 mMediaTimeTransform = xform; 4631 mMediaTimeTransformTarget = target; 4632 mMediaTimeTransformValid = true; 4633 4634 return NO_ERROR; 4635} 4636 4637#define min(a, b) ((a) < (b) ? (a) : (b)) 4638 4639// implementation of getNextBuffer for tracks whose buffers have timestamps 4640status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4641 AudioBufferProvider::Buffer* buffer, int64_t pts) 4642{ 4643 if (pts == AudioBufferProvider::kInvalidPTS) { 4644 buffer->raw = 0; 4645 buffer->frameCount = 0; 4646 mTimedAudioOutputOnTime = false; 4647 return INVALID_OPERATION; 4648 } 4649 4650 Mutex::Autolock _l(mTimedBufferQueueLock); 4651 4652 ALOG_ASSERT(!mQueueHeadInFlight, 4653 "getNextBuffer called without releaseBuffer!"); 4654 4655 while (true) { 4656 4657 // if we have no timed buffers, then fail 4658 if (mTimedBufferQueue.isEmpty()) { 4659 buffer->raw = 0; 4660 buffer->frameCount = 0; 4661 return NOT_ENOUGH_DATA; 4662 } 4663 4664 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4665 4666 // calculate the PTS of the head of the timed buffer queue expressed in 4667 // local time 4668 int64_t headLocalPTS; 4669 { 4670 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4671 4672 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4673 4674 if (mMediaTimeTransform.a_to_b_denom == 0) { 4675 // the transform represents a pause, so yield silence 4676 timedYieldSilence_l(buffer->frameCount, buffer); 4677 return NO_ERROR; 4678 } 4679 4680 int64_t transformedPTS; 4681 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4682 &transformedPTS)) { 4683 // the transform failed. this shouldn't happen, but if it does 4684 // then just drop this buffer 4685 ALOGW("timedGetNextBuffer transform failed"); 4686 buffer->raw = 0; 4687 buffer->frameCount = 0; 4688 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4689 return NO_ERROR; 4690 } 4691 4692 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4693 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4694 &headLocalPTS)) { 4695 buffer->raw = 0; 4696 buffer->frameCount = 0; 4697 return INVALID_OPERATION; 4698 } 4699 } else { 4700 headLocalPTS = transformedPTS; 4701 } 4702 } 4703 4704 // adjust the head buffer's PTS to reflect the portion of the head buffer 4705 // that has already been consumed 4706 int64_t effectivePTS = headLocalPTS + 4707 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4708 4709 // Calculate the delta in samples between the head of the input buffer 4710 // queue and the start of the next output buffer that will be written. 4711 // If the transformation fails because of over or underflow, it means 4712 // that the sample's position in the output stream is so far out of 4713 // whack that it should just be dropped. 4714 int64_t sampleDelta; 4715 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4716 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4717 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4718 " mix"); 4719 continue; 4720 } 4721 if (!mLocalTimeToSampleTransform.doForwardTransform( 4722 (effectivePTS - pts) << 32, &sampleDelta)) { 4723 ALOGV("*** too late during sample rate transform: dropped buffer"); 4724 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 4725 continue; 4726 } 4727 4728 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 4729 " sampleDelta=[%d.%08x]", 4730 head.pts(), head.position(), pts, 4731 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 4732 + (sampleDelta >> 32)), 4733 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4734 4735 // if the delta between the ideal placement for the next input sample and 4736 // the current output position is within this threshold, then we will 4737 // concatenate the next input samples to the previous output 4738 const int64_t kSampleContinuityThreshold = 4739 (static_cast<int64_t>(sampleRate()) << 32) / 250; 4740 4741 // if this is the first buffer of audio that we're emitting from this track 4742 // then it should be almost exactly on time. 4743 const int64_t kSampleStartupThreshold = 1LL << 32; 4744 4745 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4746 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4747 // the next input is close enough to being on time, so concatenate it 4748 // with the last output 4749 timedYieldSamples_l(buffer); 4750 4751 ALOGVV("*** on time: head.pos=%d frameCount=%u", 4752 head.position(), buffer->frameCount); 4753 return NO_ERROR; 4754 } 4755 4756 // Looks like our output is not on time. Reset our on timed status. 4757 // Next time we mix samples from our input queue, then should be within 4758 // the StartupThreshold. 4759 mTimedAudioOutputOnTime = false; 4760 if (sampleDelta > 0) { 4761 // the gap between the current output position and the proper start of 4762 // the next input sample is too big, so fill it with silence 4763 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4764 4765 timedYieldSilence_l(framesUntilNextInput, buffer); 4766 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4767 return NO_ERROR; 4768 } else { 4769 // the next input sample is late 4770 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4771 size_t onTimeSamplePosition = 4772 head.position() + lateFrames * mCblk->frameSize; 4773 4774 if (onTimeSamplePosition > head.buffer()->size()) { 4775 // all the remaining samples in the head are too late, so 4776 // drop it and move on 4777 ALOGV("*** too late: dropped buffer"); 4778 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 4779 continue; 4780 } else { 4781 // skip over the late samples 4782 head.setPosition(onTimeSamplePosition); 4783 4784 // yield the available samples 4785 timedYieldSamples_l(buffer); 4786 4787 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4788 return NO_ERROR; 4789 } 4790 } 4791 } 4792} 4793 4794// Yield samples from the timed buffer queue head up to the given output 4795// buffer's capacity. 4796// 4797// Caller must hold mTimedBufferQueueLock 4798void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 4799 AudioBufferProvider::Buffer* buffer) { 4800 4801 const TimedBuffer& head = mTimedBufferQueue[0]; 4802 4803 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4804 head.position()); 4805 4806 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4807 mCblk->frameSize); 4808 size_t framesRequested = buffer->frameCount; 4809 buffer->frameCount = min(framesLeftInHead, framesRequested); 4810 4811 mQueueHeadInFlight = true; 4812 mTimedAudioOutputOnTime = true; 4813} 4814 4815// Yield samples of silence up to the given output buffer's capacity 4816// 4817// Caller must hold mTimedBufferQueueLock 4818void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 4819 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4820 4821 // lazily allocate a buffer filled with silence 4822 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4823 delete [] mTimedSilenceBuffer; 4824 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4825 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4826 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4827 } 4828 4829 buffer->raw = mTimedSilenceBuffer; 4830 size_t framesRequested = buffer->frameCount; 4831 buffer->frameCount = min(numFrames, framesRequested); 4832 4833 mTimedAudioOutputOnTime = false; 4834} 4835 4836// AudioBufferProvider interface 4837void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4838 AudioBufferProvider::Buffer* buffer) { 4839 4840 Mutex::Autolock _l(mTimedBufferQueueLock); 4841 4842 // If the buffer which was just released is part of the buffer at the head 4843 // of the queue, be sure to update the amt of the buffer which has been 4844 // consumed. If the buffer being returned is not part of the head of the 4845 // queue, its either because the buffer is part of the silence buffer, or 4846 // because the head of the timed queue was trimmed after the mixer called 4847 // getNextBuffer but before the mixer called releaseBuffer. 4848 if (buffer->raw == mTimedSilenceBuffer) { 4849 ALOG_ASSERT(!mQueueHeadInFlight, 4850 "Queue head in flight during release of silence buffer!"); 4851 goto done; 4852 } 4853 4854 ALOG_ASSERT(mQueueHeadInFlight, 4855 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 4856 " head in flight."); 4857 4858 if (mTimedBufferQueue.size()) { 4859 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4860 4861 void* start = head.buffer()->pointer(); 4862 void* end = reinterpret_cast<void*>( 4863 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 4864 + head.buffer()->size()); 4865 4866 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 4867 "released buffer not within the head of the timed buffer" 4868 " queue; qHead = [%p, %p], released buffer = %p", 4869 start, end, buffer->raw); 4870 4871 head.setPosition(head.position() + 4872 (buffer->frameCount * mCblk->frameSize)); 4873 mQueueHeadInFlight = false; 4874 4875 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 4876 "Bad bookkeeping during releaseBuffer! Should have at" 4877 " least %u queued frames, but we think we have only %u", 4878 buffer->frameCount, mFramesPendingInQueue); 4879 4880 mFramesPendingInQueue -= buffer->frameCount; 4881 4882 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 4883 || mTrimQueueHeadOnRelease) { 4884 trimTimedBufferQueueHead_l("releaseBuffer"); 4885 mTrimQueueHeadOnRelease = false; 4886 } 4887 } else { 4888 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 4889 " buffers in the timed buffer queue"); 4890 } 4891 4892done: 4893 buffer->raw = 0; 4894 buffer->frameCount = 0; 4895} 4896 4897uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4898 Mutex::Autolock _l(mTimedBufferQueueLock); 4899 return mFramesPendingInQueue; 4900} 4901 4902AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4903 : mPTS(0), mPosition(0) {} 4904 4905AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4906 const sp<IMemory>& buffer, int64_t pts) 4907 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4908 4909// ---------------------------------------------------------------------------- 4910 4911// RecordTrack constructor must be called with AudioFlinger::mLock held 4912AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4913 RecordThread *thread, 4914 const sp<Client>& client, 4915 uint32_t sampleRate, 4916 audio_format_t format, 4917 uint32_t channelMask, 4918 int frameCount, 4919 int sessionId) 4920 : TrackBase(thread, client, sampleRate, format, 4921 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4922 mOverflow(false) 4923{ 4924 if (mCblk != NULL) { 4925 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4926 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4927 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4928 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4929 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4930 } else { 4931 mCblk->frameSize = sizeof(int8_t); 4932 } 4933 } 4934} 4935 4936AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4937{ 4938 sp<ThreadBase> thread = mThread.promote(); 4939 if (thread != 0) { 4940 AudioSystem::releaseInput(thread->id()); 4941 } 4942} 4943 4944// AudioBufferProvider interface 4945status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4946{ 4947 audio_track_cblk_t* cblk = this->cblk(); 4948 uint32_t framesAvail; 4949 uint32_t framesReq = buffer->frameCount; 4950 4951 // Check if last stepServer failed, try to step now 4952 if (mStepServerFailed) { 4953 if (!step()) goto getNextBuffer_exit; 4954 ALOGV("stepServer recovered"); 4955 mStepServerFailed = false; 4956 } 4957 4958 framesAvail = cblk->framesAvailable_l(); 4959 4960 if (CC_LIKELY(framesAvail)) { 4961 uint32_t s = cblk->server; 4962 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4963 4964 if (framesReq > framesAvail) { 4965 framesReq = framesAvail; 4966 } 4967 if (framesReq > bufferEnd - s) { 4968 framesReq = bufferEnd - s; 4969 } 4970 4971 buffer->raw = getBuffer(s, framesReq); 4972 if (buffer->raw == NULL) goto getNextBuffer_exit; 4973 4974 buffer->frameCount = framesReq; 4975 return NO_ERROR; 4976 } 4977 4978getNextBuffer_exit: 4979 buffer->raw = NULL; 4980 buffer->frameCount = 0; 4981 return NOT_ENOUGH_DATA; 4982} 4983 4984status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 4985 int triggerSession) 4986{ 4987 sp<ThreadBase> thread = mThread.promote(); 4988 if (thread != 0) { 4989 RecordThread *recordThread = (RecordThread *)thread.get(); 4990 return recordThread->start(this, event, triggerSession); 4991 } else { 4992 return BAD_VALUE; 4993 } 4994} 4995 4996void AudioFlinger::RecordThread::RecordTrack::stop() 4997{ 4998 sp<ThreadBase> thread = mThread.promote(); 4999 if (thread != 0) { 5000 RecordThread *recordThread = (RecordThread *)thread.get(); 5001 recordThread->stop(this); 5002 TrackBase::reset(); 5003 // Force overrun condition to avoid false overrun callback until first data is 5004 // read from buffer 5005 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5006 } 5007} 5008 5009void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5010{ 5011 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5012 (mClient == 0) ? getpid_cached : mClient->pid(), 5013 mFormat, 5014 mChannelMask, 5015 mSessionId, 5016 mFrameCount, 5017 mState, 5018 mCblk->sampleRate, 5019 mCblk->server, 5020 mCblk->user); 5021} 5022 5023 5024// ---------------------------------------------------------------------------- 5025 5026AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5027 PlaybackThread *playbackThread, 5028 DuplicatingThread *sourceThread, 5029 uint32_t sampleRate, 5030 audio_format_t format, 5031 uint32_t channelMask, 5032 int frameCount) 5033 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5034 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5035 mActive(false), mSourceThread(sourceThread) 5036{ 5037 5038 if (mCblk != NULL) { 5039 mCblk->flags |= CBLK_DIRECTION_OUT; 5040 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5041 mOutBuffer.frameCount = 0; 5042 playbackThread->mTracks.add(this); 5043 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5044 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5045 mCblk, mBuffer, mCblk->buffers, 5046 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5047 } else { 5048 ALOGW("Error creating output track on thread %p", playbackThread); 5049 } 5050} 5051 5052AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5053{ 5054 clearBufferQueue(); 5055} 5056 5057status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5058 int triggerSession) 5059{ 5060 status_t status = Track::start(event, triggerSession); 5061 if (status != NO_ERROR) { 5062 return status; 5063 } 5064 5065 mActive = true; 5066 mRetryCount = 127; 5067 return status; 5068} 5069 5070void AudioFlinger::PlaybackThread::OutputTrack::stop() 5071{ 5072 Track::stop(); 5073 clearBufferQueue(); 5074 mOutBuffer.frameCount = 0; 5075 mActive = false; 5076} 5077 5078bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5079{ 5080 Buffer *pInBuffer; 5081 Buffer inBuffer; 5082 uint32_t channelCount = mChannelCount; 5083 bool outputBufferFull = false; 5084 inBuffer.frameCount = frames; 5085 inBuffer.i16 = data; 5086 5087 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5088 5089 if (!mActive && frames != 0) { 5090 start(); 5091 sp<ThreadBase> thread = mThread.promote(); 5092 if (thread != 0) { 5093 MixerThread *mixerThread = (MixerThread *)thread.get(); 5094 if (mCblk->frameCount > frames){ 5095 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5096 uint32_t startFrames = (mCblk->frameCount - frames); 5097 pInBuffer = new Buffer; 5098 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5099 pInBuffer->frameCount = startFrames; 5100 pInBuffer->i16 = pInBuffer->mBuffer; 5101 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5102 mBufferQueue.add(pInBuffer); 5103 } else { 5104 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5105 } 5106 } 5107 } 5108 } 5109 5110 while (waitTimeLeftMs) { 5111 // First write pending buffers, then new data 5112 if (mBufferQueue.size()) { 5113 pInBuffer = mBufferQueue.itemAt(0); 5114 } else { 5115 pInBuffer = &inBuffer; 5116 } 5117 5118 if (pInBuffer->frameCount == 0) { 5119 break; 5120 } 5121 5122 if (mOutBuffer.frameCount == 0) { 5123 mOutBuffer.frameCount = pInBuffer->frameCount; 5124 nsecs_t startTime = systemTime(); 5125 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5126 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5127 outputBufferFull = true; 5128 break; 5129 } 5130 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5131 if (waitTimeLeftMs >= waitTimeMs) { 5132 waitTimeLeftMs -= waitTimeMs; 5133 } else { 5134 waitTimeLeftMs = 0; 5135 } 5136 } 5137 5138 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5139 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5140 mCblk->stepUser(outFrames); 5141 pInBuffer->frameCount -= outFrames; 5142 pInBuffer->i16 += outFrames * channelCount; 5143 mOutBuffer.frameCount -= outFrames; 5144 mOutBuffer.i16 += outFrames * channelCount; 5145 5146 if (pInBuffer->frameCount == 0) { 5147 if (mBufferQueue.size()) { 5148 mBufferQueue.removeAt(0); 5149 delete [] pInBuffer->mBuffer; 5150 delete pInBuffer; 5151 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5152 } else { 5153 break; 5154 } 5155 } 5156 } 5157 5158 // If we could not write all frames, allocate a buffer and queue it for next time. 5159 if (inBuffer.frameCount) { 5160 sp<ThreadBase> thread = mThread.promote(); 5161 if (thread != 0 && !thread->standby()) { 5162 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5163 pInBuffer = new Buffer; 5164 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5165 pInBuffer->frameCount = inBuffer.frameCount; 5166 pInBuffer->i16 = pInBuffer->mBuffer; 5167 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5168 mBufferQueue.add(pInBuffer); 5169 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5170 } else { 5171 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5172 } 5173 } 5174 } 5175 5176 // Calling write() with a 0 length buffer, means that no more data will be written: 5177 // If no more buffers are pending, fill output track buffer to make sure it is started 5178 // by output mixer. 5179 if (frames == 0 && mBufferQueue.size() == 0) { 5180 if (mCblk->user < mCblk->frameCount) { 5181 frames = mCblk->frameCount - mCblk->user; 5182 pInBuffer = new Buffer; 5183 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5184 pInBuffer->frameCount = frames; 5185 pInBuffer->i16 = pInBuffer->mBuffer; 5186 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5187 mBufferQueue.add(pInBuffer); 5188 } else if (mActive) { 5189 stop(); 5190 } 5191 } 5192 5193 return outputBufferFull; 5194} 5195 5196status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5197{ 5198 int active; 5199 status_t result; 5200 audio_track_cblk_t* cblk = mCblk; 5201 uint32_t framesReq = buffer->frameCount; 5202 5203// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5204 buffer->frameCount = 0; 5205 5206 uint32_t framesAvail = cblk->framesAvailable(); 5207 5208 5209 if (framesAvail == 0) { 5210 Mutex::Autolock _l(cblk->lock); 5211 goto start_loop_here; 5212 while (framesAvail == 0) { 5213 active = mActive; 5214 if (CC_UNLIKELY(!active)) { 5215 ALOGV("Not active and NO_MORE_BUFFERS"); 5216 return NO_MORE_BUFFERS; 5217 } 5218 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5219 if (result != NO_ERROR) { 5220 return NO_MORE_BUFFERS; 5221 } 5222 // read the server count again 5223 start_loop_here: 5224 framesAvail = cblk->framesAvailable_l(); 5225 } 5226 } 5227 5228// if (framesAvail < framesReq) { 5229// return NO_MORE_BUFFERS; 5230// } 5231 5232 if (framesReq > framesAvail) { 5233 framesReq = framesAvail; 5234 } 5235 5236 uint32_t u = cblk->user; 5237 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5238 5239 if (framesReq > bufferEnd - u) { 5240 framesReq = bufferEnd - u; 5241 } 5242 5243 buffer->frameCount = framesReq; 5244 buffer->raw = (void *)cblk->buffer(u); 5245 return NO_ERROR; 5246} 5247 5248 5249void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5250{ 5251 size_t size = mBufferQueue.size(); 5252 5253 for (size_t i = 0; i < size; i++) { 5254 Buffer *pBuffer = mBufferQueue.itemAt(i); 5255 delete [] pBuffer->mBuffer; 5256 delete pBuffer; 5257 } 5258 mBufferQueue.clear(); 5259} 5260 5261// ---------------------------------------------------------------------------- 5262 5263AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5264 : RefBase(), 5265 mAudioFlinger(audioFlinger), 5266 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5267 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5268 mPid(pid), 5269 mTimedTrackCount(0) 5270{ 5271 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5272} 5273 5274// Client destructor must be called with AudioFlinger::mLock held 5275AudioFlinger::Client::~Client() 5276{ 5277 mAudioFlinger->removeClient_l(mPid); 5278} 5279 5280sp<MemoryDealer> AudioFlinger::Client::heap() const 5281{ 5282 return mMemoryDealer; 5283} 5284 5285// Reserve one of the limited slots for a timed audio track associated 5286// with this client 5287bool AudioFlinger::Client::reserveTimedTrack() 5288{ 5289 const int kMaxTimedTracksPerClient = 4; 5290 5291 Mutex::Autolock _l(mTimedTrackLock); 5292 5293 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5294 ALOGW("can not create timed track - pid %d has exceeded the limit", 5295 mPid); 5296 return false; 5297 } 5298 5299 mTimedTrackCount++; 5300 return true; 5301} 5302 5303// Release a slot for a timed audio track 5304void AudioFlinger::Client::releaseTimedTrack() 5305{ 5306 Mutex::Autolock _l(mTimedTrackLock); 5307 mTimedTrackCount--; 5308} 5309 5310// ---------------------------------------------------------------------------- 5311 5312AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5313 const sp<IAudioFlingerClient>& client, 5314 pid_t pid) 5315 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5316{ 5317} 5318 5319AudioFlinger::NotificationClient::~NotificationClient() 5320{ 5321} 5322 5323void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5324{ 5325 sp<NotificationClient> keep(this); 5326 mAudioFlinger->removeNotificationClient(mPid); 5327} 5328 5329// ---------------------------------------------------------------------------- 5330 5331AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5332 : BnAudioTrack(), 5333 mTrack(track) 5334{ 5335} 5336 5337AudioFlinger::TrackHandle::~TrackHandle() { 5338 // just stop the track on deletion, associated resources 5339 // will be freed from the main thread once all pending buffers have 5340 // been played. Unless it's not in the active track list, in which 5341 // case we free everything now... 5342 mTrack->destroy(); 5343} 5344 5345sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5346 return mTrack->getCblk(); 5347} 5348 5349status_t AudioFlinger::TrackHandle::start() { 5350 return mTrack->start(); 5351} 5352 5353void AudioFlinger::TrackHandle::stop() { 5354 mTrack->stop(); 5355} 5356 5357void AudioFlinger::TrackHandle::flush() { 5358 mTrack->flush(); 5359} 5360 5361void AudioFlinger::TrackHandle::mute(bool e) { 5362 mTrack->mute(e); 5363} 5364 5365void AudioFlinger::TrackHandle::pause() { 5366 mTrack->pause(); 5367} 5368 5369status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5370{ 5371 return mTrack->attachAuxEffect(EffectId); 5372} 5373 5374status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5375 sp<IMemory>* buffer) { 5376 if (!mTrack->isTimedTrack()) 5377 return INVALID_OPERATION; 5378 5379 PlaybackThread::TimedTrack* tt = 5380 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5381 return tt->allocateTimedBuffer(size, buffer); 5382} 5383 5384status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5385 int64_t pts) { 5386 if (!mTrack->isTimedTrack()) 5387 return INVALID_OPERATION; 5388 5389 PlaybackThread::TimedTrack* tt = 5390 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5391 return tt->queueTimedBuffer(buffer, pts); 5392} 5393 5394status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5395 const LinearTransform& xform, int target) { 5396 5397 if (!mTrack->isTimedTrack()) 5398 return INVALID_OPERATION; 5399 5400 PlaybackThread::TimedTrack* tt = 5401 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5402 return tt->setMediaTimeTransform( 5403 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5404} 5405 5406status_t AudioFlinger::TrackHandle::onTransact( 5407 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5408{ 5409 return BnAudioTrack::onTransact(code, data, reply, flags); 5410} 5411 5412// ---------------------------------------------------------------------------- 5413 5414sp<IAudioRecord> AudioFlinger::openRecord( 5415 pid_t pid, 5416 audio_io_handle_t input, 5417 uint32_t sampleRate, 5418 audio_format_t format, 5419 uint32_t channelMask, 5420 int frameCount, 5421 IAudioFlinger::track_flags_t flags, 5422 int *sessionId, 5423 status_t *status) 5424{ 5425 sp<RecordThread::RecordTrack> recordTrack; 5426 sp<RecordHandle> recordHandle; 5427 sp<Client> client; 5428 status_t lStatus; 5429 RecordThread *thread; 5430 size_t inFrameCount; 5431 int lSessionId; 5432 5433 // check calling permissions 5434 if (!recordingAllowed()) { 5435 lStatus = PERMISSION_DENIED; 5436 goto Exit; 5437 } 5438 5439 // add client to list 5440 { // scope for mLock 5441 Mutex::Autolock _l(mLock); 5442 thread = checkRecordThread_l(input); 5443 if (thread == NULL) { 5444 lStatus = BAD_VALUE; 5445 goto Exit; 5446 } 5447 5448 client = registerPid_l(pid); 5449 5450 // If no audio session id is provided, create one here 5451 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5452 lSessionId = *sessionId; 5453 } else { 5454 lSessionId = nextUniqueId(); 5455 if (sessionId != NULL) { 5456 *sessionId = lSessionId; 5457 } 5458 } 5459 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5460 recordTrack = thread->createRecordTrack_l(client, 5461 sampleRate, 5462 format, 5463 channelMask, 5464 frameCount, 5465 lSessionId, 5466 &lStatus); 5467 } 5468 if (lStatus != NO_ERROR) { 5469 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5470 // destructor is called by the TrackBase destructor with mLock held 5471 client.clear(); 5472 recordTrack.clear(); 5473 goto Exit; 5474 } 5475 5476 // return to handle to client 5477 recordHandle = new RecordHandle(recordTrack); 5478 lStatus = NO_ERROR; 5479 5480Exit: 5481 if (status) { 5482 *status = lStatus; 5483 } 5484 return recordHandle; 5485} 5486 5487// ---------------------------------------------------------------------------- 5488 5489AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5490 : BnAudioRecord(), 5491 mRecordTrack(recordTrack) 5492{ 5493} 5494 5495AudioFlinger::RecordHandle::~RecordHandle() { 5496 stop(); 5497} 5498 5499sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5500 return mRecordTrack->getCblk(); 5501} 5502 5503status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5504 ALOGV("RecordHandle::start()"); 5505 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5506} 5507 5508void AudioFlinger::RecordHandle::stop() { 5509 ALOGV("RecordHandle::stop()"); 5510 mRecordTrack->stop(); 5511} 5512 5513status_t AudioFlinger::RecordHandle::onTransact( 5514 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5515{ 5516 return BnAudioRecord::onTransact(code, data, reply, flags); 5517} 5518 5519// ---------------------------------------------------------------------------- 5520 5521AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5522 AudioStreamIn *input, 5523 uint32_t sampleRate, 5524 uint32_t channels, 5525 audio_io_handle_t id, 5526 uint32_t device) : 5527 ThreadBase(audioFlinger, id, device, RECORD), 5528 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5529 // mRsmpInIndex and mInputBytes set by readInputParameters() 5530 mReqChannelCount(popcount(channels)), 5531 mReqSampleRate(sampleRate) 5532 // mBytesRead is only meaningful while active, and so is cleared in start() 5533 // (but might be better to also clear here for dump?) 5534{ 5535 snprintf(mName, kNameLength, "AudioIn_%X", id); 5536 5537 readInputParameters(); 5538} 5539 5540 5541AudioFlinger::RecordThread::~RecordThread() 5542{ 5543 delete[] mRsmpInBuffer; 5544 delete mResampler; 5545 delete[] mRsmpOutBuffer; 5546} 5547 5548void AudioFlinger::RecordThread::onFirstRef() 5549{ 5550 run(mName, PRIORITY_URGENT_AUDIO); 5551} 5552 5553status_t AudioFlinger::RecordThread::readyToRun() 5554{ 5555 status_t status = initCheck(); 5556 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5557 return status; 5558} 5559 5560bool AudioFlinger::RecordThread::threadLoop() 5561{ 5562 AudioBufferProvider::Buffer buffer; 5563 sp<RecordTrack> activeTrack; 5564 Vector< sp<EffectChain> > effectChains; 5565 5566 nsecs_t lastWarning = 0; 5567 5568 acquireWakeLock(); 5569 5570 // start recording 5571 while (!exitPending()) { 5572 5573 processConfigEvents(); 5574 5575 { // scope for mLock 5576 Mutex::Autolock _l(mLock); 5577 checkForNewParameters_l(); 5578 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5579 if (!mStandby) { 5580 mInput->stream->common.standby(&mInput->stream->common); 5581 mStandby = true; 5582 } 5583 5584 if (exitPending()) break; 5585 5586 releaseWakeLock_l(); 5587 ALOGV("RecordThread: loop stopping"); 5588 // go to sleep 5589 mWaitWorkCV.wait(mLock); 5590 ALOGV("RecordThread: loop starting"); 5591 acquireWakeLock_l(); 5592 continue; 5593 } 5594 if (mActiveTrack != 0) { 5595 if (mActiveTrack->mState == TrackBase::PAUSING) { 5596 if (!mStandby) { 5597 mInput->stream->common.standby(&mInput->stream->common); 5598 mStandby = true; 5599 } 5600 mActiveTrack.clear(); 5601 mStartStopCond.broadcast(); 5602 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5603 if (mReqChannelCount != mActiveTrack->channelCount()) { 5604 mActiveTrack.clear(); 5605 mStartStopCond.broadcast(); 5606 } else if (mBytesRead != 0) { 5607 // record start succeeds only if first read from audio input 5608 // succeeds 5609 if (mBytesRead > 0) { 5610 mActiveTrack->mState = TrackBase::ACTIVE; 5611 } else { 5612 mActiveTrack.clear(); 5613 } 5614 mStartStopCond.broadcast(); 5615 } 5616 mStandby = false; 5617 } 5618 } 5619 lockEffectChains_l(effectChains); 5620 } 5621 5622 if (mActiveTrack != 0) { 5623 if (mActiveTrack->mState != TrackBase::ACTIVE && 5624 mActiveTrack->mState != TrackBase::RESUMING) { 5625 unlockEffectChains(effectChains); 5626 usleep(kRecordThreadSleepUs); 5627 continue; 5628 } 5629 for (size_t i = 0; i < effectChains.size(); i ++) { 5630 effectChains[i]->process_l(); 5631 } 5632 5633 buffer.frameCount = mFrameCount; 5634 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5635 size_t framesOut = buffer.frameCount; 5636 if (mResampler == NULL) { 5637 // no resampling 5638 while (framesOut) { 5639 size_t framesIn = mFrameCount - mRsmpInIndex; 5640 if (framesIn) { 5641 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5642 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5643 if (framesIn > framesOut) 5644 framesIn = framesOut; 5645 mRsmpInIndex += framesIn; 5646 framesOut -= framesIn; 5647 if ((int)mChannelCount == mReqChannelCount || 5648 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5649 memcpy(dst, src, framesIn * mFrameSize); 5650 } else { 5651 int16_t *src16 = (int16_t *)src; 5652 int16_t *dst16 = (int16_t *)dst; 5653 if (mChannelCount == 1) { 5654 while (framesIn--) { 5655 *dst16++ = *src16; 5656 *dst16++ = *src16++; 5657 } 5658 } else { 5659 while (framesIn--) { 5660 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5661 src16 += 2; 5662 } 5663 } 5664 } 5665 } 5666 if (framesOut && mFrameCount == mRsmpInIndex) { 5667 if (framesOut == mFrameCount && 5668 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5669 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5670 framesOut = 0; 5671 } else { 5672 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5673 mRsmpInIndex = 0; 5674 } 5675 if (mBytesRead < 0) { 5676 ALOGE("Error reading audio input"); 5677 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5678 // Force input into standby so that it tries to 5679 // recover at next read attempt 5680 mInput->stream->common.standby(&mInput->stream->common); 5681 usleep(kRecordThreadSleepUs); 5682 } 5683 mRsmpInIndex = mFrameCount; 5684 framesOut = 0; 5685 buffer.frameCount = 0; 5686 } 5687 } 5688 } 5689 } else { 5690 // resampling 5691 5692 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5693 // alter output frame count as if we were expecting stereo samples 5694 if (mChannelCount == 1 && mReqChannelCount == 1) { 5695 framesOut >>= 1; 5696 } 5697 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5698 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5699 // are 32 bit aligned which should be always true. 5700 if (mChannelCount == 2 && mReqChannelCount == 1) { 5701 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5702 // the resampler always outputs stereo samples: do post stereo to mono conversion 5703 int16_t *src = (int16_t *)mRsmpOutBuffer; 5704 int16_t *dst = buffer.i16; 5705 while (framesOut--) { 5706 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5707 src += 2; 5708 } 5709 } else { 5710 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5711 } 5712 5713 } 5714 if (mFramestoDrop == 0) { 5715 mActiveTrack->releaseBuffer(&buffer); 5716 } else { 5717 if (mFramestoDrop > 0) { 5718 mFramestoDrop -= buffer.frameCount; 5719 if (mFramestoDrop < 0) { 5720 mFramestoDrop = 0; 5721 } 5722 } 5723 } 5724 mActiveTrack->overflow(); 5725 } 5726 // client isn't retrieving buffers fast enough 5727 else { 5728 if (!mActiveTrack->setOverflow()) { 5729 nsecs_t now = systemTime(); 5730 if ((now - lastWarning) > kWarningThrottleNs) { 5731 ALOGW("RecordThread: buffer overflow"); 5732 lastWarning = now; 5733 } 5734 } 5735 // Release the processor for a while before asking for a new buffer. 5736 // This will give the application more chance to read from the buffer and 5737 // clear the overflow. 5738 usleep(kRecordThreadSleepUs); 5739 } 5740 } 5741 // enable changes in effect chain 5742 unlockEffectChains(effectChains); 5743 effectChains.clear(); 5744 } 5745 5746 if (!mStandby) { 5747 mInput->stream->common.standby(&mInput->stream->common); 5748 } 5749 mActiveTrack.clear(); 5750 5751 mStartStopCond.broadcast(); 5752 5753 releaseWakeLock(); 5754 5755 ALOGV("RecordThread %p exiting", this); 5756 return false; 5757} 5758 5759 5760sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5761 const sp<AudioFlinger::Client>& client, 5762 uint32_t sampleRate, 5763 audio_format_t format, 5764 int channelMask, 5765 int frameCount, 5766 int sessionId, 5767 status_t *status) 5768{ 5769 sp<RecordTrack> track; 5770 status_t lStatus; 5771 5772 lStatus = initCheck(); 5773 if (lStatus != NO_ERROR) { 5774 ALOGE("Audio driver not initialized."); 5775 goto Exit; 5776 } 5777 5778 { // scope for mLock 5779 Mutex::Autolock _l(mLock); 5780 5781 track = new RecordTrack(this, client, sampleRate, 5782 format, channelMask, frameCount, sessionId); 5783 5784 if (track->getCblk() == 0) { 5785 lStatus = NO_MEMORY; 5786 goto Exit; 5787 } 5788 5789 mTrack = track.get(); 5790 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5791 bool suspend = audio_is_bluetooth_sco_device( 5792 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5793 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5794 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5795 } 5796 lStatus = NO_ERROR; 5797 5798Exit: 5799 if (status) { 5800 *status = lStatus; 5801 } 5802 return track; 5803} 5804 5805status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5806 AudioSystem::sync_event_t event, 5807 int triggerSession) 5808{ 5809 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5810 sp<ThreadBase> strongMe = this; 5811 status_t status = NO_ERROR; 5812 5813 if (event == AudioSystem::SYNC_EVENT_NONE) { 5814 mSyncStartEvent.clear(); 5815 mFramestoDrop = 0; 5816 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5817 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5818 triggerSession, 5819 recordTrack->sessionId(), 5820 syncStartEventCallback, 5821 this); 5822 mFramestoDrop = -1; 5823 } 5824 5825 { 5826 AutoMutex lock(mLock); 5827 if (mActiveTrack != 0) { 5828 if (recordTrack != mActiveTrack.get()) { 5829 status = -EBUSY; 5830 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5831 mActiveTrack->mState = TrackBase::ACTIVE; 5832 } 5833 return status; 5834 } 5835 5836 recordTrack->mState = TrackBase::IDLE; 5837 mActiveTrack = recordTrack; 5838 mLock.unlock(); 5839 status_t status = AudioSystem::startInput(mId); 5840 mLock.lock(); 5841 if (status != NO_ERROR) { 5842 mActiveTrack.clear(); 5843 clearSyncStartEvent(); 5844 return status; 5845 } 5846 mRsmpInIndex = mFrameCount; 5847 mBytesRead = 0; 5848 if (mResampler != NULL) { 5849 mResampler->reset(); 5850 } 5851 mActiveTrack->mState = TrackBase::RESUMING; 5852 // signal thread to start 5853 ALOGV("Signal record thread"); 5854 mWaitWorkCV.signal(); 5855 // do not wait for mStartStopCond if exiting 5856 if (exitPending()) { 5857 mActiveTrack.clear(); 5858 status = INVALID_OPERATION; 5859 goto startError; 5860 } 5861 mStartStopCond.wait(mLock); 5862 if (mActiveTrack == 0) { 5863 ALOGV("Record failed to start"); 5864 status = BAD_VALUE; 5865 goto startError; 5866 } 5867 ALOGV("Record started OK"); 5868 return status; 5869 } 5870startError: 5871 AudioSystem::stopInput(mId); 5872 clearSyncStartEvent(); 5873 return status; 5874} 5875 5876void AudioFlinger::RecordThread::clearSyncStartEvent() 5877{ 5878 if (mSyncStartEvent != 0) { 5879 mSyncStartEvent->cancel(); 5880 } 5881 mSyncStartEvent.clear(); 5882} 5883 5884void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5885{ 5886 sp<SyncEvent> strongEvent = event.promote(); 5887 5888 if (strongEvent != 0) { 5889 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5890 me->handleSyncStartEvent(strongEvent); 5891 } 5892} 5893 5894void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5895{ 5896 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5897 mActiveTrack.get(), 5898 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5899 event->listenerSession()); 5900 5901 if (mActiveTrack != 0 && 5902 event == mSyncStartEvent) { 5903 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5904 // from audio HAL 5905 mFramestoDrop = mFrameCount * 2; 5906 mSyncStartEvent.clear(); 5907 } 5908} 5909 5910void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5911 ALOGV("RecordThread::stop"); 5912 sp<ThreadBase> strongMe = this; 5913 { 5914 AutoMutex lock(mLock); 5915 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5916 mActiveTrack->mState = TrackBase::PAUSING; 5917 // do not wait for mStartStopCond if exiting 5918 if (exitPending()) { 5919 return; 5920 } 5921 mStartStopCond.wait(mLock); 5922 // if we have been restarted, recordTrack == mActiveTrack.get() here 5923 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5924 mLock.unlock(); 5925 AudioSystem::stopInput(mId); 5926 mLock.lock(); 5927 ALOGV("Record stopped OK"); 5928 } 5929 } 5930 } 5931} 5932 5933bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5934{ 5935 return false; 5936} 5937 5938status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5939{ 5940 if (!isValidSyncEvent(event)) { 5941 return BAD_VALUE; 5942 } 5943 5944 Mutex::Autolock _l(mLock); 5945 5946 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5947 mTrack->setSyncEvent(event); 5948 return NO_ERROR; 5949 } 5950 return NAME_NOT_FOUND; 5951} 5952 5953status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5954{ 5955 const size_t SIZE = 256; 5956 char buffer[SIZE]; 5957 String8 result; 5958 5959 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5960 result.append(buffer); 5961 5962 if (mActiveTrack != 0) { 5963 result.append("Active Track:\n"); 5964 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5965 mActiveTrack->dump(buffer, SIZE); 5966 result.append(buffer); 5967 5968 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5969 result.append(buffer); 5970 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5971 result.append(buffer); 5972 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5973 result.append(buffer); 5974 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5975 result.append(buffer); 5976 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5977 result.append(buffer); 5978 5979 5980 } else { 5981 result.append("No record client\n"); 5982 } 5983 write(fd, result.string(), result.size()); 5984 5985 dumpBase(fd, args); 5986 dumpEffectChains(fd, args); 5987 5988 return NO_ERROR; 5989} 5990 5991// AudioBufferProvider interface 5992status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5993{ 5994 size_t framesReq = buffer->frameCount; 5995 size_t framesReady = mFrameCount - mRsmpInIndex; 5996 int channelCount; 5997 5998 if (framesReady == 0) { 5999 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6000 if (mBytesRead < 0) { 6001 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6002 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6003 // Force input into standby so that it tries to 6004 // recover at next read attempt 6005 mInput->stream->common.standby(&mInput->stream->common); 6006 usleep(kRecordThreadSleepUs); 6007 } 6008 buffer->raw = NULL; 6009 buffer->frameCount = 0; 6010 return NOT_ENOUGH_DATA; 6011 } 6012 mRsmpInIndex = 0; 6013 framesReady = mFrameCount; 6014 } 6015 6016 if (framesReq > framesReady) { 6017 framesReq = framesReady; 6018 } 6019 6020 if (mChannelCount == 1 && mReqChannelCount == 2) { 6021 channelCount = 1; 6022 } else { 6023 channelCount = 2; 6024 } 6025 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6026 buffer->frameCount = framesReq; 6027 return NO_ERROR; 6028} 6029 6030// AudioBufferProvider interface 6031void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6032{ 6033 mRsmpInIndex += buffer->frameCount; 6034 buffer->frameCount = 0; 6035} 6036 6037bool AudioFlinger::RecordThread::checkForNewParameters_l() 6038{ 6039 bool reconfig = false; 6040 6041 while (!mNewParameters.isEmpty()) { 6042 status_t status = NO_ERROR; 6043 String8 keyValuePair = mNewParameters[0]; 6044 AudioParameter param = AudioParameter(keyValuePair); 6045 int value; 6046 audio_format_t reqFormat = mFormat; 6047 int reqSamplingRate = mReqSampleRate; 6048 int reqChannelCount = mReqChannelCount; 6049 6050 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6051 reqSamplingRate = value; 6052 reconfig = true; 6053 } 6054 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6055 reqFormat = (audio_format_t) value; 6056 reconfig = true; 6057 } 6058 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6059 reqChannelCount = popcount(value); 6060 reconfig = true; 6061 } 6062 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6063 // do not accept frame count changes if tracks are open as the track buffer 6064 // size depends on frame count and correct behavior would not be guaranteed 6065 // if frame count is changed after track creation 6066 if (mActiveTrack != 0) { 6067 status = INVALID_OPERATION; 6068 } else { 6069 reconfig = true; 6070 } 6071 } 6072 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6073 // forward device change to effects that have requested to be 6074 // aware of attached audio device. 6075 for (size_t i = 0; i < mEffectChains.size(); i++) { 6076 mEffectChains[i]->setDevice_l(value); 6077 } 6078 // store input device and output device but do not forward output device to audio HAL. 6079 // Note that status is ignored by the caller for output device 6080 // (see AudioFlinger::setParameters() 6081 if (value & AUDIO_DEVICE_OUT_ALL) { 6082 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6083 status = BAD_VALUE; 6084 } else { 6085 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6086 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6087 if (mTrack != NULL) { 6088 bool suspend = audio_is_bluetooth_sco_device( 6089 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6090 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6091 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6092 } 6093 } 6094 mDevice |= (uint32_t)value; 6095 } 6096 if (status == NO_ERROR) { 6097 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6098 if (status == INVALID_OPERATION) { 6099 mInput->stream->common.standby(&mInput->stream->common); 6100 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6101 keyValuePair.string()); 6102 } 6103 if (reconfig) { 6104 if (status == BAD_VALUE && 6105 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6106 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6107 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6108 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6109 (reqChannelCount <= FCC_2)) { 6110 status = NO_ERROR; 6111 } 6112 if (status == NO_ERROR) { 6113 readInputParameters(); 6114 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6115 } 6116 } 6117 } 6118 6119 mNewParameters.removeAt(0); 6120 6121 mParamStatus = status; 6122 mParamCond.signal(); 6123 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6124 // already timed out waiting for the status and will never signal the condition. 6125 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6126 } 6127 return reconfig; 6128} 6129 6130String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6131{ 6132 char *s; 6133 String8 out_s8 = String8(); 6134 6135 Mutex::Autolock _l(mLock); 6136 if (initCheck() != NO_ERROR) { 6137 return out_s8; 6138 } 6139 6140 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6141 out_s8 = String8(s); 6142 free(s); 6143 return out_s8; 6144} 6145 6146void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6147 AudioSystem::OutputDescriptor desc; 6148 void *param2 = NULL; 6149 6150 switch (event) { 6151 case AudioSystem::INPUT_OPENED: 6152 case AudioSystem::INPUT_CONFIG_CHANGED: 6153 desc.channels = mChannelMask; 6154 desc.samplingRate = mSampleRate; 6155 desc.format = mFormat; 6156 desc.frameCount = mFrameCount; 6157 desc.latency = 0; 6158 param2 = &desc; 6159 break; 6160 6161 case AudioSystem::INPUT_CLOSED: 6162 default: 6163 break; 6164 } 6165 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6166} 6167 6168void AudioFlinger::RecordThread::readInputParameters() 6169{ 6170 delete mRsmpInBuffer; 6171 // mRsmpInBuffer is always assigned a new[] below 6172 delete mRsmpOutBuffer; 6173 mRsmpOutBuffer = NULL; 6174 delete mResampler; 6175 mResampler = NULL; 6176 6177 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6178 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6179 mChannelCount = (uint16_t)popcount(mChannelMask); 6180 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6181 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6182 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6183 mFrameCount = mInputBytes / mFrameSize; 6184 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6185 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6186 6187 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6188 { 6189 int channelCount; 6190 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6191 // stereo to mono post process as the resampler always outputs stereo. 6192 if (mChannelCount == 1 && mReqChannelCount == 2) { 6193 channelCount = 1; 6194 } else { 6195 channelCount = 2; 6196 } 6197 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6198 mResampler->setSampleRate(mSampleRate); 6199 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6200 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6201 6202 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6203 if (mChannelCount == 1 && mReqChannelCount == 1) { 6204 mFrameCount >>= 1; 6205 } 6206 6207 } 6208 mRsmpInIndex = mFrameCount; 6209} 6210 6211unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6212{ 6213 Mutex::Autolock _l(mLock); 6214 if (initCheck() != NO_ERROR) { 6215 return 0; 6216 } 6217 6218 return mInput->stream->get_input_frames_lost(mInput->stream); 6219} 6220 6221uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6222{ 6223 Mutex::Autolock _l(mLock); 6224 uint32_t result = 0; 6225 if (getEffectChain_l(sessionId) != 0) { 6226 result = EFFECT_SESSION; 6227 } 6228 6229 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6230 result |= TRACK_SESSION; 6231 } 6232 6233 return result; 6234} 6235 6236AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6237{ 6238 Mutex::Autolock _l(mLock); 6239 return mTrack; 6240} 6241 6242AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6243{ 6244 Mutex::Autolock _l(mLock); 6245 return mInput; 6246} 6247 6248AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6249{ 6250 Mutex::Autolock _l(mLock); 6251 AudioStreamIn *input = mInput; 6252 mInput = NULL; 6253 return input; 6254} 6255 6256// this method must always be called either with ThreadBase mLock held or inside the thread loop 6257audio_stream_t* AudioFlinger::RecordThread::stream() const 6258{ 6259 if (mInput == NULL) { 6260 return NULL; 6261 } 6262 return &mInput->stream->common; 6263} 6264 6265 6266// ---------------------------------------------------------------------------- 6267 6268audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6269{ 6270 if (!settingsAllowed()) { 6271 return 0; 6272 } 6273 Mutex::Autolock _l(mLock); 6274 return loadHwModule_l(name); 6275} 6276 6277// loadHwModule_l() must be called with AudioFlinger::mLock held 6278audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6279{ 6280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6281 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6282 ALOGW("loadHwModule() module %s already loaded", name); 6283 return mAudioHwDevs.keyAt(i); 6284 } 6285 } 6286 6287 audio_hw_device_t *dev; 6288 6289 int rc = load_audio_interface(name, &dev); 6290 if (rc) { 6291 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6292 return 0; 6293 } 6294 6295 mHardwareStatus = AUDIO_HW_INIT; 6296 rc = dev->init_check(dev); 6297 mHardwareStatus = AUDIO_HW_IDLE; 6298 if (rc) { 6299 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6300 return 0; 6301 } 6302 6303 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6304 (NULL != dev->set_master_volume)) { 6305 AutoMutex lock(mHardwareLock); 6306 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6307 dev->set_master_volume(dev, mMasterVolume); 6308 mHardwareStatus = AUDIO_HW_IDLE; 6309 } 6310 6311 audio_module_handle_t handle = nextUniqueId(); 6312 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6313 6314 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6315 name, dev->common.module->name, dev->common.module->id, handle); 6316 6317 return handle; 6318 6319} 6320 6321audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6322 audio_devices_t *pDevices, 6323 uint32_t *pSamplingRate, 6324 audio_format_t *pFormat, 6325 audio_channel_mask_t *pChannelMask, 6326 uint32_t *pLatencyMs, 6327 audio_output_flags_t flags) 6328{ 6329 status_t status; 6330 PlaybackThread *thread = NULL; 6331 struct audio_config config = { 6332 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6333 channel_mask: pChannelMask ? *pChannelMask : 0, 6334 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6335 }; 6336 audio_stream_out_t *outStream = NULL; 6337 audio_hw_device_t *outHwDev; 6338 6339 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6340 module, 6341 (pDevices != NULL) ? (int)*pDevices : 0, 6342 config.sample_rate, 6343 config.format, 6344 config.channel_mask, 6345 flags); 6346 6347 if (pDevices == NULL || *pDevices == 0) { 6348 return 0; 6349 } 6350 6351 Mutex::Autolock _l(mLock); 6352 6353 outHwDev = findSuitableHwDev_l(module, *pDevices); 6354 if (outHwDev == NULL) 6355 return 0; 6356 6357 audio_io_handle_t id = nextUniqueId(); 6358 6359 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6360 6361 status = outHwDev->open_output_stream(outHwDev, 6362 id, 6363 *pDevices, 6364 (audio_output_flags_t)flags, 6365 &config, 6366 &outStream); 6367 6368 mHardwareStatus = AUDIO_HW_IDLE; 6369 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6370 outStream, 6371 config.sample_rate, 6372 config.format, 6373 config.channel_mask, 6374 status); 6375 6376 if (status == NO_ERROR && outStream != NULL) { 6377 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6378 6379 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6380 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6381 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6382 thread = new DirectOutputThread(this, output, id, *pDevices); 6383 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6384 } else { 6385 thread = new MixerThread(this, output, id, *pDevices); 6386 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6387 } 6388 mPlaybackThreads.add(id, thread); 6389 6390 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6391 if (pFormat != NULL) *pFormat = config.format; 6392 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6393 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6394 6395 // notify client processes of the new output creation 6396 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6397 6398 // the first primary output opened designates the primary hw device 6399 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6400 ALOGI("Using module %d has the primary audio interface", module); 6401 mPrimaryHardwareDev = outHwDev; 6402 6403 AutoMutex lock(mHardwareLock); 6404 mHardwareStatus = AUDIO_HW_SET_MODE; 6405 outHwDev->set_mode(outHwDev, mMode); 6406 6407 // Determine the level of master volume support the primary audio HAL has, 6408 // and set the initial master volume at the same time. 6409 float initialVolume = 1.0; 6410 mMasterVolumeSupportLvl = MVS_NONE; 6411 6412 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6413 if ((NULL != outHwDev->get_master_volume) && 6414 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6415 mMasterVolumeSupportLvl = MVS_FULL; 6416 } else { 6417 mMasterVolumeSupportLvl = MVS_SETONLY; 6418 initialVolume = 1.0; 6419 } 6420 6421 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6422 if ((NULL == outHwDev->set_master_volume) || 6423 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6424 mMasterVolumeSupportLvl = MVS_NONE; 6425 } 6426 // now that we have a primary device, initialize master volume on other devices 6427 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6428 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6429 6430 if ((dev != mPrimaryHardwareDev) && 6431 (NULL != dev->set_master_volume)) { 6432 dev->set_master_volume(dev, initialVolume); 6433 } 6434 } 6435 mHardwareStatus = AUDIO_HW_IDLE; 6436 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6437 ? initialVolume 6438 : 1.0; 6439 mMasterVolume = initialVolume; 6440 } 6441 return id; 6442 } 6443 6444 return 0; 6445} 6446 6447audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6448 audio_io_handle_t output2) 6449{ 6450 Mutex::Autolock _l(mLock); 6451 MixerThread *thread1 = checkMixerThread_l(output1); 6452 MixerThread *thread2 = checkMixerThread_l(output2); 6453 6454 if (thread1 == NULL || thread2 == NULL) { 6455 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6456 return 0; 6457 } 6458 6459 audio_io_handle_t id = nextUniqueId(); 6460 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6461 thread->addOutputTrack(thread2); 6462 mPlaybackThreads.add(id, thread); 6463 // notify client processes of the new output creation 6464 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6465 return id; 6466} 6467 6468status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6469{ 6470 // keep strong reference on the playback thread so that 6471 // it is not destroyed while exit() is executed 6472 sp<PlaybackThread> thread; 6473 { 6474 Mutex::Autolock _l(mLock); 6475 thread = checkPlaybackThread_l(output); 6476 if (thread == NULL) { 6477 return BAD_VALUE; 6478 } 6479 6480 ALOGV("closeOutput() %d", output); 6481 6482 if (thread->type() == ThreadBase::MIXER) { 6483 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6484 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6485 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6486 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6487 } 6488 } 6489 } 6490 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6491 mPlaybackThreads.removeItem(output); 6492 } 6493 thread->exit(); 6494 // The thread entity (active unit of execution) is no longer running here, 6495 // but the ThreadBase container still exists. 6496 6497 if (thread->type() != ThreadBase::DUPLICATING) { 6498 AudioStreamOut *out = thread->clearOutput(); 6499 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6500 // from now on thread->mOutput is NULL 6501 out->hwDev->close_output_stream(out->hwDev, out->stream); 6502 delete out; 6503 } 6504 return NO_ERROR; 6505} 6506 6507status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6508{ 6509 Mutex::Autolock _l(mLock); 6510 PlaybackThread *thread = checkPlaybackThread_l(output); 6511 6512 if (thread == NULL) { 6513 return BAD_VALUE; 6514 } 6515 6516 ALOGV("suspendOutput() %d", output); 6517 thread->suspend(); 6518 6519 return NO_ERROR; 6520} 6521 6522status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6523{ 6524 Mutex::Autolock _l(mLock); 6525 PlaybackThread *thread = checkPlaybackThread_l(output); 6526 6527 if (thread == NULL) { 6528 return BAD_VALUE; 6529 } 6530 6531 ALOGV("restoreOutput() %d", output); 6532 6533 thread->restore(); 6534 6535 return NO_ERROR; 6536} 6537 6538audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6539 audio_devices_t *pDevices, 6540 uint32_t *pSamplingRate, 6541 audio_format_t *pFormat, 6542 uint32_t *pChannelMask) 6543{ 6544 status_t status; 6545 RecordThread *thread = NULL; 6546 struct audio_config config = { 6547 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6548 channel_mask: pChannelMask ? *pChannelMask : 0, 6549 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6550 }; 6551 uint32_t reqSamplingRate = config.sample_rate; 6552 audio_format_t reqFormat = config.format; 6553 audio_channel_mask_t reqChannels = config.channel_mask; 6554 audio_stream_in_t *inStream = NULL; 6555 audio_hw_device_t *inHwDev; 6556 6557 if (pDevices == NULL || *pDevices == 0) { 6558 return 0; 6559 } 6560 6561 Mutex::Autolock _l(mLock); 6562 6563 inHwDev = findSuitableHwDev_l(module, *pDevices); 6564 if (inHwDev == NULL) 6565 return 0; 6566 6567 audio_io_handle_t id = nextUniqueId(); 6568 6569 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6570 &inStream); 6571 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6572 inStream, 6573 config.sample_rate, 6574 config.format, 6575 config.channel_mask, 6576 status); 6577 6578 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6579 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6580 // or stereo to mono conversions on 16 bit PCM inputs. 6581 if (status == BAD_VALUE && 6582 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6583 (config.sample_rate <= 2 * reqSamplingRate) && 6584 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6585 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6586 inStream = NULL; 6587 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6588 } 6589 6590 if (status == NO_ERROR && inStream != NULL) { 6591 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6592 6593 // Start record thread 6594 // RecorThread require both input and output device indication to forward to audio 6595 // pre processing modules 6596 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6597 thread = new RecordThread(this, 6598 input, 6599 reqSamplingRate, 6600 reqChannels, 6601 id, 6602 device); 6603 mRecordThreads.add(id, thread); 6604 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6605 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6606 if (pFormat != NULL) *pFormat = config.format; 6607 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6608 6609 input->stream->common.standby(&input->stream->common); 6610 6611 // notify client processes of the new input creation 6612 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6613 return id; 6614 } 6615 6616 return 0; 6617} 6618 6619status_t AudioFlinger::closeInput(audio_io_handle_t input) 6620{ 6621 // keep strong reference on the record thread so that 6622 // it is not destroyed while exit() is executed 6623 sp<RecordThread> thread; 6624 { 6625 Mutex::Autolock _l(mLock); 6626 thread = checkRecordThread_l(input); 6627 if (thread == NULL) { 6628 return BAD_VALUE; 6629 } 6630 6631 ALOGV("closeInput() %d", input); 6632 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6633 mRecordThreads.removeItem(input); 6634 } 6635 thread->exit(); 6636 // The thread entity (active unit of execution) is no longer running here, 6637 // but the ThreadBase container still exists. 6638 6639 AudioStreamIn *in = thread->clearInput(); 6640 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6641 // from now on thread->mInput is NULL 6642 in->hwDev->close_input_stream(in->hwDev, in->stream); 6643 delete in; 6644 6645 return NO_ERROR; 6646} 6647 6648status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6649{ 6650 Mutex::Autolock _l(mLock); 6651 MixerThread *dstThread = checkMixerThread_l(output); 6652 if (dstThread == NULL) { 6653 ALOGW("setStreamOutput() bad output id %d", output); 6654 return BAD_VALUE; 6655 } 6656 6657 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6658 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6659 6660 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6661 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6662 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6663 MixerThread *srcThread = (MixerThread *)thread; 6664 srcThread->invalidateTracks(stream); 6665 } 6666 } 6667 6668 return NO_ERROR; 6669} 6670 6671 6672int AudioFlinger::newAudioSessionId() 6673{ 6674 return nextUniqueId(); 6675} 6676 6677void AudioFlinger::acquireAudioSessionId(int audioSession) 6678{ 6679 Mutex::Autolock _l(mLock); 6680 pid_t caller = IPCThreadState::self()->getCallingPid(); 6681 ALOGV("acquiring %d from %d", audioSession, caller); 6682 size_t num = mAudioSessionRefs.size(); 6683 for (size_t i = 0; i< num; i++) { 6684 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6685 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6686 ref->mCnt++; 6687 ALOGV(" incremented refcount to %d", ref->mCnt); 6688 return; 6689 } 6690 } 6691 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6692 ALOGV(" added new entry for %d", audioSession); 6693} 6694 6695void AudioFlinger::releaseAudioSessionId(int audioSession) 6696{ 6697 Mutex::Autolock _l(mLock); 6698 pid_t caller = IPCThreadState::self()->getCallingPid(); 6699 ALOGV("releasing %d from %d", audioSession, caller); 6700 size_t num = mAudioSessionRefs.size(); 6701 for (size_t i = 0; i< num; i++) { 6702 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6703 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6704 ref->mCnt--; 6705 ALOGV(" decremented refcount to %d", ref->mCnt); 6706 if (ref->mCnt == 0) { 6707 mAudioSessionRefs.removeAt(i); 6708 delete ref; 6709 purgeStaleEffects_l(); 6710 } 6711 return; 6712 } 6713 } 6714 ALOGW("session id %d not found for pid %d", audioSession, caller); 6715} 6716 6717void AudioFlinger::purgeStaleEffects_l() { 6718 6719 ALOGV("purging stale effects"); 6720 6721 Vector< sp<EffectChain> > chains; 6722 6723 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6724 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6725 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6726 sp<EffectChain> ec = t->mEffectChains[j]; 6727 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6728 chains.push(ec); 6729 } 6730 } 6731 } 6732 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6733 sp<RecordThread> t = mRecordThreads.valueAt(i); 6734 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6735 sp<EffectChain> ec = t->mEffectChains[j]; 6736 chains.push(ec); 6737 } 6738 } 6739 6740 for (size_t i = 0; i < chains.size(); i++) { 6741 sp<EffectChain> ec = chains[i]; 6742 int sessionid = ec->sessionId(); 6743 sp<ThreadBase> t = ec->mThread.promote(); 6744 if (t == 0) { 6745 continue; 6746 } 6747 size_t numsessionrefs = mAudioSessionRefs.size(); 6748 bool found = false; 6749 for (size_t k = 0; k < numsessionrefs; k++) { 6750 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6751 if (ref->mSessionid == sessionid) { 6752 ALOGV(" session %d still exists for %d with %d refs", 6753 sessionid, ref->mPid, ref->mCnt); 6754 found = true; 6755 break; 6756 } 6757 } 6758 if (!found) { 6759 // remove all effects from the chain 6760 while (ec->mEffects.size()) { 6761 sp<EffectModule> effect = ec->mEffects[0]; 6762 effect->unPin(); 6763 Mutex::Autolock _l (t->mLock); 6764 t->removeEffect_l(effect); 6765 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6766 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6767 if (handle != 0) { 6768 handle->mEffect.clear(); 6769 if (handle->mHasControl && handle->mEnabled) { 6770 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6771 } 6772 } 6773 } 6774 AudioSystem::unregisterEffect(effect->id()); 6775 } 6776 } 6777 } 6778 return; 6779} 6780 6781// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6782AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6783{ 6784 return mPlaybackThreads.valueFor(output).get(); 6785} 6786 6787// checkMixerThread_l() must be called with AudioFlinger::mLock held 6788AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6789{ 6790 PlaybackThread *thread = checkPlaybackThread_l(output); 6791 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6792} 6793 6794// checkRecordThread_l() must be called with AudioFlinger::mLock held 6795AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6796{ 6797 return mRecordThreads.valueFor(input).get(); 6798} 6799 6800uint32_t AudioFlinger::nextUniqueId() 6801{ 6802 return android_atomic_inc(&mNextUniqueId); 6803} 6804 6805AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6806{ 6807 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6808 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6809 AudioStreamOut *output = thread->getOutput(); 6810 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6811 return thread; 6812 } 6813 } 6814 return NULL; 6815} 6816 6817uint32_t AudioFlinger::primaryOutputDevice_l() const 6818{ 6819 PlaybackThread *thread = primaryPlaybackThread_l(); 6820 6821 if (thread == NULL) { 6822 return 0; 6823 } 6824 6825 return thread->device(); 6826} 6827 6828sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6829 int triggerSession, 6830 int listenerSession, 6831 sync_event_callback_t callBack, 6832 void *cookie) 6833{ 6834 Mutex::Autolock _l(mLock); 6835 6836 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6837 status_t playStatus = NAME_NOT_FOUND; 6838 status_t recStatus = NAME_NOT_FOUND; 6839 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6840 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6841 if (playStatus == NO_ERROR) { 6842 return event; 6843 } 6844 } 6845 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6846 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6847 if (recStatus == NO_ERROR) { 6848 return event; 6849 } 6850 } 6851 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6852 mPendingSyncEvents.add(event); 6853 } else { 6854 ALOGV("createSyncEvent() invalid event %d", event->type()); 6855 event.clear(); 6856 } 6857 return event; 6858} 6859 6860// ---------------------------------------------------------------------------- 6861// Effect management 6862// ---------------------------------------------------------------------------- 6863 6864 6865status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6866{ 6867 Mutex::Autolock _l(mLock); 6868 return EffectQueryNumberEffects(numEffects); 6869} 6870 6871status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6872{ 6873 Mutex::Autolock _l(mLock); 6874 return EffectQueryEffect(index, descriptor); 6875} 6876 6877status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6878 effect_descriptor_t *descriptor) const 6879{ 6880 Mutex::Autolock _l(mLock); 6881 return EffectGetDescriptor(pUuid, descriptor); 6882} 6883 6884 6885sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6886 effect_descriptor_t *pDesc, 6887 const sp<IEffectClient>& effectClient, 6888 int32_t priority, 6889 audio_io_handle_t io, 6890 int sessionId, 6891 status_t *status, 6892 int *id, 6893 int *enabled) 6894{ 6895 status_t lStatus = NO_ERROR; 6896 sp<EffectHandle> handle; 6897 effect_descriptor_t desc; 6898 6899 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6900 pid, effectClient.get(), priority, sessionId, io); 6901 6902 if (pDesc == NULL) { 6903 lStatus = BAD_VALUE; 6904 goto Exit; 6905 } 6906 6907 // check audio settings permission for global effects 6908 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6909 lStatus = PERMISSION_DENIED; 6910 goto Exit; 6911 } 6912 6913 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6914 // that can only be created by audio policy manager (running in same process) 6915 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6916 lStatus = PERMISSION_DENIED; 6917 goto Exit; 6918 } 6919 6920 if (io == 0) { 6921 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6922 // output must be specified by AudioPolicyManager when using session 6923 // AUDIO_SESSION_OUTPUT_STAGE 6924 lStatus = BAD_VALUE; 6925 goto Exit; 6926 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6927 // if the output returned by getOutputForEffect() is removed before we lock the 6928 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6929 // and we will exit safely 6930 io = AudioSystem::getOutputForEffect(&desc); 6931 } 6932 } 6933 6934 { 6935 Mutex::Autolock _l(mLock); 6936 6937 6938 if (!EffectIsNullUuid(&pDesc->uuid)) { 6939 // if uuid is specified, request effect descriptor 6940 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6941 if (lStatus < 0) { 6942 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6943 goto Exit; 6944 } 6945 } else { 6946 // if uuid is not specified, look for an available implementation 6947 // of the required type in effect factory 6948 if (EffectIsNullUuid(&pDesc->type)) { 6949 ALOGW("createEffect() no effect type"); 6950 lStatus = BAD_VALUE; 6951 goto Exit; 6952 } 6953 uint32_t numEffects = 0; 6954 effect_descriptor_t d; 6955 d.flags = 0; // prevent compiler warning 6956 bool found = false; 6957 6958 lStatus = EffectQueryNumberEffects(&numEffects); 6959 if (lStatus < 0) { 6960 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6961 goto Exit; 6962 } 6963 for (uint32_t i = 0; i < numEffects; i++) { 6964 lStatus = EffectQueryEffect(i, &desc); 6965 if (lStatus < 0) { 6966 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6967 continue; 6968 } 6969 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6970 // If matching type found save effect descriptor. If the session is 6971 // 0 and the effect is not auxiliary, continue enumeration in case 6972 // an auxiliary version of this effect type is available 6973 found = true; 6974 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6975 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6976 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6977 break; 6978 } 6979 } 6980 } 6981 if (!found) { 6982 lStatus = BAD_VALUE; 6983 ALOGW("createEffect() effect not found"); 6984 goto Exit; 6985 } 6986 // For same effect type, chose auxiliary version over insert version if 6987 // connect to output mix (Compliance to OpenSL ES) 6988 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6989 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6990 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6991 } 6992 } 6993 6994 // Do not allow auxiliary effects on a session different from 0 (output mix) 6995 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6996 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6997 lStatus = INVALID_OPERATION; 6998 goto Exit; 6999 } 7000 7001 // check recording permission for visualizer 7002 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7003 !recordingAllowed()) { 7004 lStatus = PERMISSION_DENIED; 7005 goto Exit; 7006 } 7007 7008 // return effect descriptor 7009 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7010 7011 // If output is not specified try to find a matching audio session ID in one of the 7012 // output threads. 7013 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7014 // because of code checking output when entering the function. 7015 // Note: io is never 0 when creating an effect on an input 7016 if (io == 0) { 7017 // look for the thread where the specified audio session is present 7018 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7019 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7020 io = mPlaybackThreads.keyAt(i); 7021 break; 7022 } 7023 } 7024 if (io == 0) { 7025 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7026 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7027 io = mRecordThreads.keyAt(i); 7028 break; 7029 } 7030 } 7031 } 7032 // If no output thread contains the requested session ID, default to 7033 // first output. The effect chain will be moved to the correct output 7034 // thread when a track with the same session ID is created 7035 if (io == 0 && mPlaybackThreads.size()) { 7036 io = mPlaybackThreads.keyAt(0); 7037 } 7038 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7039 } 7040 ThreadBase *thread = checkRecordThread_l(io); 7041 if (thread == NULL) { 7042 thread = checkPlaybackThread_l(io); 7043 if (thread == NULL) { 7044 ALOGE("createEffect() unknown output thread"); 7045 lStatus = BAD_VALUE; 7046 goto Exit; 7047 } 7048 } 7049 7050 sp<Client> client = registerPid_l(pid); 7051 7052 // create effect on selected output thread 7053 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7054 &desc, enabled, &lStatus); 7055 if (handle != 0 && id != NULL) { 7056 *id = handle->id(); 7057 } 7058 } 7059 7060Exit: 7061 if (status != NULL) { 7062 *status = lStatus; 7063 } 7064 return handle; 7065} 7066 7067status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7068 audio_io_handle_t dstOutput) 7069{ 7070 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7071 sessionId, srcOutput, dstOutput); 7072 Mutex::Autolock _l(mLock); 7073 if (srcOutput == dstOutput) { 7074 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7075 return NO_ERROR; 7076 } 7077 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7078 if (srcThread == NULL) { 7079 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7080 return BAD_VALUE; 7081 } 7082 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7083 if (dstThread == NULL) { 7084 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7085 return BAD_VALUE; 7086 } 7087 7088 Mutex::Autolock _dl(dstThread->mLock); 7089 Mutex::Autolock _sl(srcThread->mLock); 7090 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7091 7092 return NO_ERROR; 7093} 7094 7095// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7096status_t AudioFlinger::moveEffectChain_l(int sessionId, 7097 AudioFlinger::PlaybackThread *srcThread, 7098 AudioFlinger::PlaybackThread *dstThread, 7099 bool reRegister) 7100{ 7101 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7102 sessionId, srcThread, dstThread); 7103 7104 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7105 if (chain == 0) { 7106 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7107 sessionId, srcThread); 7108 return INVALID_OPERATION; 7109 } 7110 7111 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7112 // so that a new chain is created with correct parameters when first effect is added. This is 7113 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7114 // removed. 7115 srcThread->removeEffectChain_l(chain); 7116 7117 // transfer all effects one by one so that new effect chain is created on new thread with 7118 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7119 audio_io_handle_t dstOutput = dstThread->id(); 7120 sp<EffectChain> dstChain; 7121 uint32_t strategy = 0; // prevent compiler warning 7122 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7123 while (effect != 0) { 7124 srcThread->removeEffect_l(effect); 7125 dstThread->addEffect_l(effect); 7126 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7127 if (effect->state() == EffectModule::ACTIVE || 7128 effect->state() == EffectModule::STOPPING) { 7129 effect->start(); 7130 } 7131 // if the move request is not received from audio policy manager, the effect must be 7132 // re-registered with the new strategy and output 7133 if (dstChain == 0) { 7134 dstChain = effect->chain().promote(); 7135 if (dstChain == 0) { 7136 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7137 srcThread->addEffect_l(effect); 7138 return NO_INIT; 7139 } 7140 strategy = dstChain->strategy(); 7141 } 7142 if (reRegister) { 7143 AudioSystem::unregisterEffect(effect->id()); 7144 AudioSystem::registerEffect(&effect->desc(), 7145 dstOutput, 7146 strategy, 7147 sessionId, 7148 effect->id()); 7149 } 7150 effect = chain->getEffectFromId_l(0); 7151 } 7152 7153 return NO_ERROR; 7154} 7155 7156 7157// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7158sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7159 const sp<AudioFlinger::Client>& client, 7160 const sp<IEffectClient>& effectClient, 7161 int32_t priority, 7162 int sessionId, 7163 effect_descriptor_t *desc, 7164 int *enabled, 7165 status_t *status 7166 ) 7167{ 7168 sp<EffectModule> effect; 7169 sp<EffectHandle> handle; 7170 status_t lStatus; 7171 sp<EffectChain> chain; 7172 bool chainCreated = false; 7173 bool effectCreated = false; 7174 bool effectRegistered = false; 7175 7176 lStatus = initCheck(); 7177 if (lStatus != NO_ERROR) { 7178 ALOGW("createEffect_l() Audio driver not initialized."); 7179 goto Exit; 7180 } 7181 7182 // Do not allow effects with session ID 0 on direct output or duplicating threads 7183 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7184 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7185 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7186 desc->name, sessionId); 7187 lStatus = BAD_VALUE; 7188 goto Exit; 7189 } 7190 // Only Pre processor effects are allowed on input threads and only on input threads 7191 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7192 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7193 desc->name, desc->flags, mType); 7194 lStatus = BAD_VALUE; 7195 goto Exit; 7196 } 7197 7198 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7199 7200 { // scope for mLock 7201 Mutex::Autolock _l(mLock); 7202 7203 // check for existing effect chain with the requested audio session 7204 chain = getEffectChain_l(sessionId); 7205 if (chain == 0) { 7206 // create a new chain for this session 7207 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7208 chain = new EffectChain(this, sessionId); 7209 addEffectChain_l(chain); 7210 chain->setStrategy(getStrategyForSession_l(sessionId)); 7211 chainCreated = true; 7212 } else { 7213 effect = chain->getEffectFromDesc_l(desc); 7214 } 7215 7216 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7217 7218 if (effect == 0) { 7219 int id = mAudioFlinger->nextUniqueId(); 7220 // Check CPU and memory usage 7221 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7222 if (lStatus != NO_ERROR) { 7223 goto Exit; 7224 } 7225 effectRegistered = true; 7226 // create a new effect module if none present in the chain 7227 effect = new EffectModule(this, chain, desc, id, sessionId); 7228 lStatus = effect->status(); 7229 if (lStatus != NO_ERROR) { 7230 goto Exit; 7231 } 7232 lStatus = chain->addEffect_l(effect); 7233 if (lStatus != NO_ERROR) { 7234 goto Exit; 7235 } 7236 effectCreated = true; 7237 7238 effect->setDevice(mDevice); 7239 effect->setMode(mAudioFlinger->getMode()); 7240 } 7241 // create effect handle and connect it to effect module 7242 handle = new EffectHandle(effect, client, effectClient, priority); 7243 lStatus = effect->addHandle(handle); 7244 if (enabled != NULL) { 7245 *enabled = (int)effect->isEnabled(); 7246 } 7247 } 7248 7249Exit: 7250 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7251 Mutex::Autolock _l(mLock); 7252 if (effectCreated) { 7253 chain->removeEffect_l(effect); 7254 } 7255 if (effectRegistered) { 7256 AudioSystem::unregisterEffect(effect->id()); 7257 } 7258 if (chainCreated) { 7259 removeEffectChain_l(chain); 7260 } 7261 handle.clear(); 7262 } 7263 7264 if (status != NULL) { 7265 *status = lStatus; 7266 } 7267 return handle; 7268} 7269 7270sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7271{ 7272 sp<EffectChain> chain = getEffectChain_l(sessionId); 7273 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7274} 7275 7276// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7277// PlaybackThread::mLock held 7278status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7279{ 7280 // check for existing effect chain with the requested audio session 7281 int sessionId = effect->sessionId(); 7282 sp<EffectChain> chain = getEffectChain_l(sessionId); 7283 bool chainCreated = false; 7284 7285 if (chain == 0) { 7286 // create a new chain for this session 7287 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7288 chain = new EffectChain(this, sessionId); 7289 addEffectChain_l(chain); 7290 chain->setStrategy(getStrategyForSession_l(sessionId)); 7291 chainCreated = true; 7292 } 7293 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7294 7295 if (chain->getEffectFromId_l(effect->id()) != 0) { 7296 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7297 this, effect->desc().name, chain.get()); 7298 return BAD_VALUE; 7299 } 7300 7301 status_t status = chain->addEffect_l(effect); 7302 if (status != NO_ERROR) { 7303 if (chainCreated) { 7304 removeEffectChain_l(chain); 7305 } 7306 return status; 7307 } 7308 7309 effect->setDevice(mDevice); 7310 effect->setMode(mAudioFlinger->getMode()); 7311 return NO_ERROR; 7312} 7313 7314void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7315 7316 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7317 effect_descriptor_t desc = effect->desc(); 7318 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7319 detachAuxEffect_l(effect->id()); 7320 } 7321 7322 sp<EffectChain> chain = effect->chain().promote(); 7323 if (chain != 0) { 7324 // remove effect chain if removing last effect 7325 if (chain->removeEffect_l(effect) == 0) { 7326 removeEffectChain_l(chain); 7327 } 7328 } else { 7329 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7330 } 7331} 7332 7333void AudioFlinger::ThreadBase::lockEffectChains_l( 7334 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7335{ 7336 effectChains = mEffectChains; 7337 for (size_t i = 0; i < mEffectChains.size(); i++) { 7338 mEffectChains[i]->lock(); 7339 } 7340} 7341 7342void AudioFlinger::ThreadBase::unlockEffectChains( 7343 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7344{ 7345 for (size_t i = 0; i < effectChains.size(); i++) { 7346 effectChains[i]->unlock(); 7347 } 7348} 7349 7350sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7351{ 7352 Mutex::Autolock _l(mLock); 7353 return getEffectChain_l(sessionId); 7354} 7355 7356sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7357{ 7358 size_t size = mEffectChains.size(); 7359 for (size_t i = 0; i < size; i++) { 7360 if (mEffectChains[i]->sessionId() == sessionId) { 7361 return mEffectChains[i]; 7362 } 7363 } 7364 return 0; 7365} 7366 7367void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7368{ 7369 Mutex::Autolock _l(mLock); 7370 size_t size = mEffectChains.size(); 7371 for (size_t i = 0; i < size; i++) { 7372 mEffectChains[i]->setMode_l(mode); 7373 } 7374} 7375 7376void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7377 const wp<EffectHandle>& handle, 7378 bool unpinIfLast) { 7379 7380 Mutex::Autolock _l(mLock); 7381 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7382 // delete the effect module if removing last handle on it 7383 if (effect->removeHandle(handle) == 0) { 7384 if (!effect->isPinned() || unpinIfLast) { 7385 removeEffect_l(effect); 7386 AudioSystem::unregisterEffect(effect->id()); 7387 } 7388 } 7389} 7390 7391status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7392{ 7393 int session = chain->sessionId(); 7394 int16_t *buffer = mMixBuffer; 7395 bool ownsBuffer = false; 7396 7397 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7398 if (session > 0) { 7399 // Only one effect chain can be present in direct output thread and it uses 7400 // the mix buffer as input 7401 if (mType != DIRECT) { 7402 size_t numSamples = mNormalFrameCount * mChannelCount; 7403 buffer = new int16_t[numSamples]; 7404 memset(buffer, 0, numSamples * sizeof(int16_t)); 7405 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7406 ownsBuffer = true; 7407 } 7408 7409 // Attach all tracks with same session ID to this chain. 7410 for (size_t i = 0; i < mTracks.size(); ++i) { 7411 sp<Track> track = mTracks[i]; 7412 if (session == track->sessionId()) { 7413 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7414 track->setMainBuffer(buffer); 7415 chain->incTrackCnt(); 7416 } 7417 } 7418 7419 // indicate all active tracks in the chain 7420 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7421 sp<Track> track = mActiveTracks[i].promote(); 7422 if (track == 0) continue; 7423 if (session == track->sessionId()) { 7424 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7425 chain->incActiveTrackCnt(); 7426 } 7427 } 7428 } 7429 7430 chain->setInBuffer(buffer, ownsBuffer); 7431 chain->setOutBuffer(mMixBuffer); 7432 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7433 // chains list in order to be processed last as it contains output stage effects 7434 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7435 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7436 // after track specific effects and before output stage 7437 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7438 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7439 // Effect chain for other sessions are inserted at beginning of effect 7440 // chains list to be processed before output mix effects. Relative order between other 7441 // sessions is not important 7442 size_t size = mEffectChains.size(); 7443 size_t i = 0; 7444 for (i = 0; i < size; i++) { 7445 if (mEffectChains[i]->sessionId() < session) break; 7446 } 7447 mEffectChains.insertAt(chain, i); 7448 checkSuspendOnAddEffectChain_l(chain); 7449 7450 return NO_ERROR; 7451} 7452 7453size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7454{ 7455 int session = chain->sessionId(); 7456 7457 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7458 7459 for (size_t i = 0; i < mEffectChains.size(); i++) { 7460 if (chain == mEffectChains[i]) { 7461 mEffectChains.removeAt(i); 7462 // detach all active tracks from the chain 7463 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7464 sp<Track> track = mActiveTracks[i].promote(); 7465 if (track == 0) continue; 7466 if (session == track->sessionId()) { 7467 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7468 chain.get(), session); 7469 chain->decActiveTrackCnt(); 7470 } 7471 } 7472 7473 // detach all tracks with same session ID from this chain 7474 for (size_t i = 0; i < mTracks.size(); ++i) { 7475 sp<Track> track = mTracks[i]; 7476 if (session == track->sessionId()) { 7477 track->setMainBuffer(mMixBuffer); 7478 chain->decTrackCnt(); 7479 } 7480 } 7481 break; 7482 } 7483 } 7484 return mEffectChains.size(); 7485} 7486 7487status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7488 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7489{ 7490 Mutex::Autolock _l(mLock); 7491 return attachAuxEffect_l(track, EffectId); 7492} 7493 7494status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7495 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7496{ 7497 status_t status = NO_ERROR; 7498 7499 if (EffectId == 0) { 7500 track->setAuxBuffer(0, NULL); 7501 } else { 7502 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7503 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7504 if (effect != 0) { 7505 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7506 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7507 } else { 7508 status = INVALID_OPERATION; 7509 } 7510 } else { 7511 status = BAD_VALUE; 7512 } 7513 } 7514 return status; 7515} 7516 7517void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7518{ 7519 for (size_t i = 0; i < mTracks.size(); ++i) { 7520 sp<Track> track = mTracks[i]; 7521 if (track->auxEffectId() == effectId) { 7522 attachAuxEffect_l(track, 0); 7523 } 7524 } 7525} 7526 7527status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7528{ 7529 // only one chain per input thread 7530 if (mEffectChains.size() != 0) { 7531 return INVALID_OPERATION; 7532 } 7533 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7534 7535 chain->setInBuffer(NULL); 7536 chain->setOutBuffer(NULL); 7537 7538 checkSuspendOnAddEffectChain_l(chain); 7539 7540 mEffectChains.add(chain); 7541 7542 return NO_ERROR; 7543} 7544 7545size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7546{ 7547 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7548 ALOGW_IF(mEffectChains.size() != 1, 7549 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7550 chain.get(), mEffectChains.size(), this); 7551 if (mEffectChains.size() == 1) { 7552 mEffectChains.removeAt(0); 7553 } 7554 return 0; 7555} 7556 7557// ---------------------------------------------------------------------------- 7558// EffectModule implementation 7559// ---------------------------------------------------------------------------- 7560 7561#undef LOG_TAG 7562#define LOG_TAG "AudioFlinger::EffectModule" 7563 7564AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7565 const wp<AudioFlinger::EffectChain>& chain, 7566 effect_descriptor_t *desc, 7567 int id, 7568 int sessionId) 7569 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7570 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7571{ 7572 ALOGV("Constructor %p", this); 7573 int lStatus; 7574 if (thread == NULL) { 7575 return; 7576 } 7577 7578 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7579 7580 // create effect engine from effect factory 7581 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7582 7583 if (mStatus != NO_ERROR) { 7584 return; 7585 } 7586 lStatus = init(); 7587 if (lStatus < 0) { 7588 mStatus = lStatus; 7589 goto Error; 7590 } 7591 7592 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7593 mPinned = true; 7594 } 7595 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7596 return; 7597Error: 7598 EffectRelease(mEffectInterface); 7599 mEffectInterface = NULL; 7600 ALOGV("Constructor Error %d", mStatus); 7601} 7602 7603AudioFlinger::EffectModule::~EffectModule() 7604{ 7605 ALOGV("Destructor %p", this); 7606 if (mEffectInterface != NULL) { 7607 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7608 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7609 sp<ThreadBase> thread = mThread.promote(); 7610 if (thread != 0) { 7611 audio_stream_t *stream = thread->stream(); 7612 if (stream != NULL) { 7613 stream->remove_audio_effect(stream, mEffectInterface); 7614 } 7615 } 7616 } 7617 // release effect engine 7618 EffectRelease(mEffectInterface); 7619 } 7620} 7621 7622status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7623{ 7624 status_t status; 7625 7626 Mutex::Autolock _l(mLock); 7627 int priority = handle->priority(); 7628 size_t size = mHandles.size(); 7629 sp<EffectHandle> h; 7630 size_t i; 7631 for (i = 0; i < size; i++) { 7632 h = mHandles[i].promote(); 7633 if (h == 0) continue; 7634 if (h->priority() <= priority) break; 7635 } 7636 // if inserted in first place, move effect control from previous owner to this handle 7637 if (i == 0) { 7638 bool enabled = false; 7639 if (h != 0) { 7640 enabled = h->enabled(); 7641 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7642 } 7643 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7644 status = NO_ERROR; 7645 } else { 7646 status = ALREADY_EXISTS; 7647 } 7648 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7649 mHandles.insertAt(handle, i); 7650 return status; 7651} 7652 7653size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7654{ 7655 Mutex::Autolock _l(mLock); 7656 size_t size = mHandles.size(); 7657 size_t i; 7658 for (i = 0; i < size; i++) { 7659 if (mHandles[i] == handle) break; 7660 } 7661 if (i == size) { 7662 return size; 7663 } 7664 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7665 7666 bool enabled = false; 7667 EffectHandle *hdl = handle.unsafe_get(); 7668 if (hdl != NULL) { 7669 ALOGV("removeHandle() unsafe_get OK"); 7670 enabled = hdl->enabled(); 7671 } 7672 mHandles.removeAt(i); 7673 size = mHandles.size(); 7674 // if removed from first place, move effect control from this handle to next in line 7675 if (i == 0 && size != 0) { 7676 sp<EffectHandle> h = mHandles[0].promote(); 7677 if (h != 0) { 7678 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7679 } 7680 } 7681 7682 // Prevent calls to process() and other functions on effect interface from now on. 7683 // The effect engine will be released by the destructor when the last strong reference on 7684 // this object is released which can happen after next process is called. 7685 if (size == 0 && !mPinned) { 7686 mState = DESTROYED; 7687 } 7688 7689 return size; 7690} 7691 7692sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7693{ 7694 Mutex::Autolock _l(mLock); 7695 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7696} 7697 7698void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7699{ 7700 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7701 // keep a strong reference on this EffectModule to avoid calling the 7702 // destructor before we exit 7703 sp<EffectModule> keep(this); 7704 { 7705 sp<ThreadBase> thread = mThread.promote(); 7706 if (thread != 0) { 7707 thread->disconnectEffect(keep, handle, unpinIfLast); 7708 } 7709 } 7710} 7711 7712void AudioFlinger::EffectModule::updateState() { 7713 Mutex::Autolock _l(mLock); 7714 7715 switch (mState) { 7716 case RESTART: 7717 reset_l(); 7718 // FALL THROUGH 7719 7720 case STARTING: 7721 // clear auxiliary effect input buffer for next accumulation 7722 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7723 memset(mConfig.inputCfg.buffer.raw, 7724 0, 7725 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7726 } 7727 start_l(); 7728 mState = ACTIVE; 7729 break; 7730 case STOPPING: 7731 stop_l(); 7732 mDisableWaitCnt = mMaxDisableWaitCnt; 7733 mState = STOPPED; 7734 break; 7735 case STOPPED: 7736 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7737 // turn off sequence. 7738 if (--mDisableWaitCnt == 0) { 7739 reset_l(); 7740 mState = IDLE; 7741 } 7742 break; 7743 default: //IDLE , ACTIVE, DESTROYED 7744 break; 7745 } 7746} 7747 7748void AudioFlinger::EffectModule::process() 7749{ 7750 Mutex::Autolock _l(mLock); 7751 7752 if (mState == DESTROYED || mEffectInterface == NULL || 7753 mConfig.inputCfg.buffer.raw == NULL || 7754 mConfig.outputCfg.buffer.raw == NULL) { 7755 return; 7756 } 7757 7758 if (isProcessEnabled()) { 7759 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7760 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7761 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7762 mConfig.inputCfg.buffer.s32, 7763 mConfig.inputCfg.buffer.frameCount/2); 7764 } 7765 7766 // do the actual processing in the effect engine 7767 int ret = (*mEffectInterface)->process(mEffectInterface, 7768 &mConfig.inputCfg.buffer, 7769 &mConfig.outputCfg.buffer); 7770 7771 // force transition to IDLE state when engine is ready 7772 if (mState == STOPPED && ret == -ENODATA) { 7773 mDisableWaitCnt = 1; 7774 } 7775 7776 // clear auxiliary effect input buffer for next accumulation 7777 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7778 memset(mConfig.inputCfg.buffer.raw, 0, 7779 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7780 } 7781 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7782 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7783 // If an insert effect is idle and input buffer is different from output buffer, 7784 // accumulate input onto output 7785 sp<EffectChain> chain = mChain.promote(); 7786 if (chain != 0 && chain->activeTrackCnt() != 0) { 7787 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7788 int16_t *in = mConfig.inputCfg.buffer.s16; 7789 int16_t *out = mConfig.outputCfg.buffer.s16; 7790 for (size_t i = 0; i < frameCnt; i++) { 7791 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7792 } 7793 } 7794 } 7795} 7796 7797void AudioFlinger::EffectModule::reset_l() 7798{ 7799 if (mEffectInterface == NULL) { 7800 return; 7801 } 7802 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7803} 7804 7805status_t AudioFlinger::EffectModule::configure() 7806{ 7807 uint32_t channels; 7808 if (mEffectInterface == NULL) { 7809 return NO_INIT; 7810 } 7811 7812 sp<ThreadBase> thread = mThread.promote(); 7813 if (thread == 0) { 7814 return DEAD_OBJECT; 7815 } 7816 7817 // TODO: handle configuration of effects replacing track process 7818 if (thread->channelCount() == 1) { 7819 channels = AUDIO_CHANNEL_OUT_MONO; 7820 } else { 7821 channels = AUDIO_CHANNEL_OUT_STEREO; 7822 } 7823 7824 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7825 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7826 } else { 7827 mConfig.inputCfg.channels = channels; 7828 } 7829 mConfig.outputCfg.channels = channels; 7830 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7831 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7832 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7833 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7834 mConfig.inputCfg.bufferProvider.cookie = NULL; 7835 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7836 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7837 mConfig.outputCfg.bufferProvider.cookie = NULL; 7838 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7839 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7840 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7841 // Insert effect: 7842 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7843 // always overwrites output buffer: input buffer == output buffer 7844 // - in other sessions: 7845 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7846 // other effect: overwrites output buffer: input buffer == output buffer 7847 // Auxiliary effect: 7848 // accumulates in output buffer: input buffer != output buffer 7849 // Therefore: accumulate <=> input buffer != output buffer 7850 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7851 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7852 } else { 7853 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7854 } 7855 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7856 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7857 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7858 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7859 7860 ALOGV("configure() %p thread %p buffer %p framecount %d", 7861 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7862 7863 status_t cmdStatus; 7864 uint32_t size = sizeof(int); 7865 status_t status = (*mEffectInterface)->command(mEffectInterface, 7866 EFFECT_CMD_SET_CONFIG, 7867 sizeof(effect_config_t), 7868 &mConfig, 7869 &size, 7870 &cmdStatus); 7871 if (status == 0) { 7872 status = cmdStatus; 7873 } 7874 7875 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7876 (1000 * mConfig.outputCfg.buffer.frameCount); 7877 7878 return status; 7879} 7880 7881status_t AudioFlinger::EffectModule::init() 7882{ 7883 Mutex::Autolock _l(mLock); 7884 if (mEffectInterface == NULL) { 7885 return NO_INIT; 7886 } 7887 status_t cmdStatus; 7888 uint32_t size = sizeof(status_t); 7889 status_t status = (*mEffectInterface)->command(mEffectInterface, 7890 EFFECT_CMD_INIT, 7891 0, 7892 NULL, 7893 &size, 7894 &cmdStatus); 7895 if (status == 0) { 7896 status = cmdStatus; 7897 } 7898 return status; 7899} 7900 7901status_t AudioFlinger::EffectModule::start() 7902{ 7903 Mutex::Autolock _l(mLock); 7904 return start_l(); 7905} 7906 7907status_t AudioFlinger::EffectModule::start_l() 7908{ 7909 if (mEffectInterface == NULL) { 7910 return NO_INIT; 7911 } 7912 status_t cmdStatus; 7913 uint32_t size = sizeof(status_t); 7914 status_t status = (*mEffectInterface)->command(mEffectInterface, 7915 EFFECT_CMD_ENABLE, 7916 0, 7917 NULL, 7918 &size, 7919 &cmdStatus); 7920 if (status == 0) { 7921 status = cmdStatus; 7922 } 7923 if (status == 0 && 7924 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7925 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7926 sp<ThreadBase> thread = mThread.promote(); 7927 if (thread != 0) { 7928 audio_stream_t *stream = thread->stream(); 7929 if (stream != NULL) { 7930 stream->add_audio_effect(stream, mEffectInterface); 7931 } 7932 } 7933 } 7934 return status; 7935} 7936 7937status_t AudioFlinger::EffectModule::stop() 7938{ 7939 Mutex::Autolock _l(mLock); 7940 return stop_l(); 7941} 7942 7943status_t AudioFlinger::EffectModule::stop_l() 7944{ 7945 if (mEffectInterface == NULL) { 7946 return NO_INIT; 7947 } 7948 status_t cmdStatus; 7949 uint32_t size = sizeof(status_t); 7950 status_t status = (*mEffectInterface)->command(mEffectInterface, 7951 EFFECT_CMD_DISABLE, 7952 0, 7953 NULL, 7954 &size, 7955 &cmdStatus); 7956 if (status == 0) { 7957 status = cmdStatus; 7958 } 7959 if (status == 0 && 7960 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7961 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7962 sp<ThreadBase> thread = mThread.promote(); 7963 if (thread != 0) { 7964 audio_stream_t *stream = thread->stream(); 7965 if (stream != NULL) { 7966 stream->remove_audio_effect(stream, mEffectInterface); 7967 } 7968 } 7969 } 7970 return status; 7971} 7972 7973status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7974 uint32_t cmdSize, 7975 void *pCmdData, 7976 uint32_t *replySize, 7977 void *pReplyData) 7978{ 7979 Mutex::Autolock _l(mLock); 7980// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7981 7982 if (mState == DESTROYED || mEffectInterface == NULL) { 7983 return NO_INIT; 7984 } 7985 status_t status = (*mEffectInterface)->command(mEffectInterface, 7986 cmdCode, 7987 cmdSize, 7988 pCmdData, 7989 replySize, 7990 pReplyData); 7991 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7992 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7993 for (size_t i = 1; i < mHandles.size(); i++) { 7994 sp<EffectHandle> h = mHandles[i].promote(); 7995 if (h != 0) { 7996 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7997 } 7998 } 7999 } 8000 return status; 8001} 8002 8003status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8004{ 8005 8006 Mutex::Autolock _l(mLock); 8007 ALOGV("setEnabled %p enabled %d", this, enabled); 8008 8009 if (enabled != isEnabled()) { 8010 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8011 if (enabled && status != NO_ERROR) { 8012 return status; 8013 } 8014 8015 switch (mState) { 8016 // going from disabled to enabled 8017 case IDLE: 8018 mState = STARTING; 8019 break; 8020 case STOPPED: 8021 mState = RESTART; 8022 break; 8023 case STOPPING: 8024 mState = ACTIVE; 8025 break; 8026 8027 // going from enabled to disabled 8028 case RESTART: 8029 mState = STOPPED; 8030 break; 8031 case STARTING: 8032 mState = IDLE; 8033 break; 8034 case ACTIVE: 8035 mState = STOPPING; 8036 break; 8037 case DESTROYED: 8038 return NO_ERROR; // simply ignore as we are being destroyed 8039 } 8040 for (size_t i = 1; i < mHandles.size(); i++) { 8041 sp<EffectHandle> h = mHandles[i].promote(); 8042 if (h != 0) { 8043 h->setEnabled(enabled); 8044 } 8045 } 8046 } 8047 return NO_ERROR; 8048} 8049 8050bool AudioFlinger::EffectModule::isEnabled() const 8051{ 8052 switch (mState) { 8053 case RESTART: 8054 case STARTING: 8055 case ACTIVE: 8056 return true; 8057 case IDLE: 8058 case STOPPING: 8059 case STOPPED: 8060 case DESTROYED: 8061 default: 8062 return false; 8063 } 8064} 8065 8066bool AudioFlinger::EffectModule::isProcessEnabled() const 8067{ 8068 switch (mState) { 8069 case RESTART: 8070 case ACTIVE: 8071 case STOPPING: 8072 case STOPPED: 8073 return true; 8074 case IDLE: 8075 case STARTING: 8076 case DESTROYED: 8077 default: 8078 return false; 8079 } 8080} 8081 8082status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8083{ 8084 Mutex::Autolock _l(mLock); 8085 status_t status = NO_ERROR; 8086 8087 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8088 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8089 if (isProcessEnabled() && 8090 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8091 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8092 status_t cmdStatus; 8093 uint32_t volume[2]; 8094 uint32_t *pVolume = NULL; 8095 uint32_t size = sizeof(volume); 8096 volume[0] = *left; 8097 volume[1] = *right; 8098 if (controller) { 8099 pVolume = volume; 8100 } 8101 status = (*mEffectInterface)->command(mEffectInterface, 8102 EFFECT_CMD_SET_VOLUME, 8103 size, 8104 volume, 8105 &size, 8106 pVolume); 8107 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8108 *left = volume[0]; 8109 *right = volume[1]; 8110 } 8111 } 8112 return status; 8113} 8114 8115status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8116{ 8117 Mutex::Autolock _l(mLock); 8118 status_t status = NO_ERROR; 8119 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8120 // audio pre processing modules on RecordThread can receive both output and 8121 // input device indication in the same call 8122 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8123 if (dev) { 8124 status_t cmdStatus; 8125 uint32_t size = sizeof(status_t); 8126 8127 status = (*mEffectInterface)->command(mEffectInterface, 8128 EFFECT_CMD_SET_DEVICE, 8129 sizeof(uint32_t), 8130 &dev, 8131 &size, 8132 &cmdStatus); 8133 if (status == NO_ERROR) { 8134 status = cmdStatus; 8135 } 8136 } 8137 dev = device & AUDIO_DEVICE_IN_ALL; 8138 if (dev) { 8139 status_t cmdStatus; 8140 uint32_t size = sizeof(status_t); 8141 8142 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8143 EFFECT_CMD_SET_INPUT_DEVICE, 8144 sizeof(uint32_t), 8145 &dev, 8146 &size, 8147 &cmdStatus); 8148 if (status2 == NO_ERROR) { 8149 status2 = cmdStatus; 8150 } 8151 if (status == NO_ERROR) { 8152 status = status2; 8153 } 8154 } 8155 } 8156 return status; 8157} 8158 8159status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8160{ 8161 Mutex::Autolock _l(mLock); 8162 status_t status = NO_ERROR; 8163 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8164 status_t cmdStatus; 8165 uint32_t size = sizeof(status_t); 8166 status = (*mEffectInterface)->command(mEffectInterface, 8167 EFFECT_CMD_SET_AUDIO_MODE, 8168 sizeof(audio_mode_t), 8169 &mode, 8170 &size, 8171 &cmdStatus); 8172 if (status == NO_ERROR) { 8173 status = cmdStatus; 8174 } 8175 } 8176 return status; 8177} 8178 8179void AudioFlinger::EffectModule::setSuspended(bool suspended) 8180{ 8181 Mutex::Autolock _l(mLock); 8182 mSuspended = suspended; 8183} 8184 8185bool AudioFlinger::EffectModule::suspended() const 8186{ 8187 Mutex::Autolock _l(mLock); 8188 return mSuspended; 8189} 8190 8191status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8192{ 8193 const size_t SIZE = 256; 8194 char buffer[SIZE]; 8195 String8 result; 8196 8197 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8198 result.append(buffer); 8199 8200 bool locked = tryLock(mLock); 8201 // failed to lock - AudioFlinger is probably deadlocked 8202 if (!locked) { 8203 result.append("\t\tCould not lock Fx mutex:\n"); 8204 } 8205 8206 result.append("\t\tSession Status State Engine:\n"); 8207 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8208 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8209 result.append(buffer); 8210 8211 result.append("\t\tDescriptor:\n"); 8212 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8213 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8214 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8215 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8216 result.append(buffer); 8217 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8218 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8219 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8220 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8221 result.append(buffer); 8222 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8223 mDescriptor.apiVersion, 8224 mDescriptor.flags); 8225 result.append(buffer); 8226 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8227 mDescriptor.name); 8228 result.append(buffer); 8229 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8230 mDescriptor.implementor); 8231 result.append(buffer); 8232 8233 result.append("\t\t- Input configuration:\n"); 8234 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8235 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8236 (uint32_t)mConfig.inputCfg.buffer.raw, 8237 mConfig.inputCfg.buffer.frameCount, 8238 mConfig.inputCfg.samplingRate, 8239 mConfig.inputCfg.channels, 8240 mConfig.inputCfg.format); 8241 result.append(buffer); 8242 8243 result.append("\t\t- Output configuration:\n"); 8244 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8245 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8246 (uint32_t)mConfig.outputCfg.buffer.raw, 8247 mConfig.outputCfg.buffer.frameCount, 8248 mConfig.outputCfg.samplingRate, 8249 mConfig.outputCfg.channels, 8250 mConfig.outputCfg.format); 8251 result.append(buffer); 8252 8253 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8254 result.append(buffer); 8255 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8256 for (size_t i = 0; i < mHandles.size(); ++i) { 8257 sp<EffectHandle> handle = mHandles[i].promote(); 8258 if (handle != 0) { 8259 handle->dump(buffer, SIZE); 8260 result.append(buffer); 8261 } 8262 } 8263 8264 result.append("\n"); 8265 8266 write(fd, result.string(), result.length()); 8267 8268 if (locked) { 8269 mLock.unlock(); 8270 } 8271 8272 return NO_ERROR; 8273} 8274 8275// ---------------------------------------------------------------------------- 8276// EffectHandle implementation 8277// ---------------------------------------------------------------------------- 8278 8279#undef LOG_TAG 8280#define LOG_TAG "AudioFlinger::EffectHandle" 8281 8282AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8283 const sp<AudioFlinger::Client>& client, 8284 const sp<IEffectClient>& effectClient, 8285 int32_t priority) 8286 : BnEffect(), 8287 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8288 mPriority(priority), mHasControl(false), mEnabled(false) 8289{ 8290 ALOGV("constructor %p", this); 8291 8292 if (client == 0) { 8293 return; 8294 } 8295 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8296 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8297 if (mCblkMemory != 0) { 8298 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8299 8300 if (mCblk != NULL) { 8301 new(mCblk) effect_param_cblk_t(); 8302 mBuffer = (uint8_t *)mCblk + bufOffset; 8303 } 8304 } else { 8305 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8306 return; 8307 } 8308} 8309 8310AudioFlinger::EffectHandle::~EffectHandle() 8311{ 8312 ALOGV("Destructor %p", this); 8313 disconnect(false); 8314 ALOGV("Destructor DONE %p", this); 8315} 8316 8317status_t AudioFlinger::EffectHandle::enable() 8318{ 8319 ALOGV("enable %p", this); 8320 if (!mHasControl) return INVALID_OPERATION; 8321 if (mEffect == 0) return DEAD_OBJECT; 8322 8323 if (mEnabled) { 8324 return NO_ERROR; 8325 } 8326 8327 mEnabled = true; 8328 8329 sp<ThreadBase> thread = mEffect->thread().promote(); 8330 if (thread != 0) { 8331 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8332 } 8333 8334 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8335 if (mEffect->suspended()) { 8336 return NO_ERROR; 8337 } 8338 8339 status_t status = mEffect->setEnabled(true); 8340 if (status != NO_ERROR) { 8341 if (thread != 0) { 8342 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8343 } 8344 mEnabled = false; 8345 } 8346 return status; 8347} 8348 8349status_t AudioFlinger::EffectHandle::disable() 8350{ 8351 ALOGV("disable %p", this); 8352 if (!mHasControl) return INVALID_OPERATION; 8353 if (mEffect == 0) return DEAD_OBJECT; 8354 8355 if (!mEnabled) { 8356 return NO_ERROR; 8357 } 8358 mEnabled = false; 8359 8360 if (mEffect->suspended()) { 8361 return NO_ERROR; 8362 } 8363 8364 status_t status = mEffect->setEnabled(false); 8365 8366 sp<ThreadBase> thread = mEffect->thread().promote(); 8367 if (thread != 0) { 8368 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8369 } 8370 8371 return status; 8372} 8373 8374void AudioFlinger::EffectHandle::disconnect() 8375{ 8376 disconnect(true); 8377} 8378 8379void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8380{ 8381 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8382 if (mEffect == 0) { 8383 return; 8384 } 8385 mEffect->disconnect(this, unpinIfLast); 8386 8387 if (mHasControl && mEnabled) { 8388 sp<ThreadBase> thread = mEffect->thread().promote(); 8389 if (thread != 0) { 8390 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8391 } 8392 } 8393 8394 // release sp on module => module destructor can be called now 8395 mEffect.clear(); 8396 if (mClient != 0) { 8397 if (mCblk != NULL) { 8398 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8399 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8400 } 8401 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8402 // Client destructor must run with AudioFlinger mutex locked 8403 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8404 mClient.clear(); 8405 } 8406} 8407 8408status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8409 uint32_t cmdSize, 8410 void *pCmdData, 8411 uint32_t *replySize, 8412 void *pReplyData) 8413{ 8414// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8415// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8416 8417 // only get parameter command is permitted for applications not controlling the effect 8418 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8419 return INVALID_OPERATION; 8420 } 8421 if (mEffect == 0) return DEAD_OBJECT; 8422 if (mClient == 0) return INVALID_OPERATION; 8423 8424 // handle commands that are not forwarded transparently to effect engine 8425 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8426 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8427 // no risk to block the whole media server process or mixer threads is we are stuck here 8428 Mutex::Autolock _l(mCblk->lock); 8429 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8430 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8431 mCblk->serverIndex = 0; 8432 mCblk->clientIndex = 0; 8433 return BAD_VALUE; 8434 } 8435 status_t status = NO_ERROR; 8436 while (mCblk->serverIndex < mCblk->clientIndex) { 8437 int reply; 8438 uint32_t rsize = sizeof(int); 8439 int *p = (int *)(mBuffer + mCblk->serverIndex); 8440 int size = *p++; 8441 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8442 ALOGW("command(): invalid parameter block size"); 8443 break; 8444 } 8445 effect_param_t *param = (effect_param_t *)p; 8446 if (param->psize == 0 || param->vsize == 0) { 8447 ALOGW("command(): null parameter or value size"); 8448 mCblk->serverIndex += size; 8449 continue; 8450 } 8451 uint32_t psize = sizeof(effect_param_t) + 8452 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8453 param->vsize; 8454 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8455 psize, 8456 p, 8457 &rsize, 8458 &reply); 8459 // stop at first error encountered 8460 if (ret != NO_ERROR) { 8461 status = ret; 8462 *(int *)pReplyData = reply; 8463 break; 8464 } else if (reply != NO_ERROR) { 8465 *(int *)pReplyData = reply; 8466 break; 8467 } 8468 mCblk->serverIndex += size; 8469 } 8470 mCblk->serverIndex = 0; 8471 mCblk->clientIndex = 0; 8472 return status; 8473 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8474 *(int *)pReplyData = NO_ERROR; 8475 return enable(); 8476 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8477 *(int *)pReplyData = NO_ERROR; 8478 return disable(); 8479 } 8480 8481 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8482} 8483 8484void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8485{ 8486 ALOGV("setControl %p control %d", this, hasControl); 8487 8488 mHasControl = hasControl; 8489 mEnabled = enabled; 8490 8491 if (signal && mEffectClient != 0) { 8492 mEffectClient->controlStatusChanged(hasControl); 8493 } 8494} 8495 8496void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8497 uint32_t cmdSize, 8498 void *pCmdData, 8499 uint32_t replySize, 8500 void *pReplyData) 8501{ 8502 if (mEffectClient != 0) { 8503 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8504 } 8505} 8506 8507 8508 8509void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8510{ 8511 if (mEffectClient != 0) { 8512 mEffectClient->enableStatusChanged(enabled); 8513 } 8514} 8515 8516status_t AudioFlinger::EffectHandle::onTransact( 8517 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8518{ 8519 return BnEffect::onTransact(code, data, reply, flags); 8520} 8521 8522 8523void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8524{ 8525 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8526 8527 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8528 (mClient == 0) ? getpid_cached : mClient->pid(), 8529 mPriority, 8530 mHasControl, 8531 !locked, 8532 mCblk ? mCblk->clientIndex : 0, 8533 mCblk ? mCblk->serverIndex : 0 8534 ); 8535 8536 if (locked) { 8537 mCblk->lock.unlock(); 8538 } 8539} 8540 8541#undef LOG_TAG 8542#define LOG_TAG "AudioFlinger::EffectChain" 8543 8544AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8545 int sessionId) 8546 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8547 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8548 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8549{ 8550 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8551 if (thread == NULL) { 8552 return; 8553 } 8554 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8555 thread->frameCount(); 8556} 8557 8558AudioFlinger::EffectChain::~EffectChain() 8559{ 8560 if (mOwnInBuffer) { 8561 delete mInBuffer; 8562 } 8563 8564} 8565 8566// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8567sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8568{ 8569 size_t size = mEffects.size(); 8570 8571 for (size_t i = 0; i < size; i++) { 8572 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8573 return mEffects[i]; 8574 } 8575 } 8576 return 0; 8577} 8578 8579// getEffectFromId_l() must be called with ThreadBase::mLock held 8580sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8581{ 8582 size_t size = mEffects.size(); 8583 8584 for (size_t i = 0; i < size; i++) { 8585 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8586 if (id == 0 || mEffects[i]->id() == id) { 8587 return mEffects[i]; 8588 } 8589 } 8590 return 0; 8591} 8592 8593// getEffectFromType_l() must be called with ThreadBase::mLock held 8594sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8595 const effect_uuid_t *type) 8596{ 8597 size_t size = mEffects.size(); 8598 8599 for (size_t i = 0; i < size; i++) { 8600 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8601 return mEffects[i]; 8602 } 8603 } 8604 return 0; 8605} 8606 8607// Must be called with EffectChain::mLock locked 8608void AudioFlinger::EffectChain::process_l() 8609{ 8610 sp<ThreadBase> thread = mThread.promote(); 8611 if (thread == 0) { 8612 ALOGW("process_l(): cannot promote mixer thread"); 8613 return; 8614 } 8615 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8616 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8617 // always process effects unless no more tracks are on the session and the effect tail 8618 // has been rendered 8619 bool doProcess = true; 8620 if (!isGlobalSession) { 8621 bool tracksOnSession = (trackCnt() != 0); 8622 8623 if (!tracksOnSession && mTailBufferCount == 0) { 8624 doProcess = false; 8625 } 8626 8627 if (activeTrackCnt() == 0) { 8628 // if no track is active and the effect tail has not been rendered, 8629 // the input buffer must be cleared here as the mixer process will not do it 8630 if (tracksOnSession || mTailBufferCount > 0) { 8631 size_t numSamples = thread->frameCount() * thread->channelCount(); 8632 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8633 if (mTailBufferCount > 0) { 8634 mTailBufferCount--; 8635 } 8636 } 8637 } 8638 } 8639 8640 size_t size = mEffects.size(); 8641 if (doProcess) { 8642 for (size_t i = 0; i < size; i++) { 8643 mEffects[i]->process(); 8644 } 8645 } 8646 for (size_t i = 0; i < size; i++) { 8647 mEffects[i]->updateState(); 8648 } 8649} 8650 8651// addEffect_l() must be called with PlaybackThread::mLock held 8652status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8653{ 8654 effect_descriptor_t desc = effect->desc(); 8655 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8656 8657 Mutex::Autolock _l(mLock); 8658 effect->setChain(this); 8659 sp<ThreadBase> thread = mThread.promote(); 8660 if (thread == 0) { 8661 return NO_INIT; 8662 } 8663 effect->setThread(thread); 8664 8665 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8666 // Auxiliary effects are inserted at the beginning of mEffects vector as 8667 // they are processed first and accumulated in chain input buffer 8668 mEffects.insertAt(effect, 0); 8669 8670 // the input buffer for auxiliary effect contains mono samples in 8671 // 32 bit format. This is to avoid saturation in AudoMixer 8672 // accumulation stage. Saturation is done in EffectModule::process() before 8673 // calling the process in effect engine 8674 size_t numSamples = thread->frameCount(); 8675 int32_t *buffer = new int32_t[numSamples]; 8676 memset(buffer, 0, numSamples * sizeof(int32_t)); 8677 effect->setInBuffer((int16_t *)buffer); 8678 // auxiliary effects output samples to chain input buffer for further processing 8679 // by insert effects 8680 effect->setOutBuffer(mInBuffer); 8681 } else { 8682 // Insert effects are inserted at the end of mEffects vector as they are processed 8683 // after track and auxiliary effects. 8684 // Insert effect order as a function of indicated preference: 8685 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8686 // another effect is present 8687 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8688 // last effect claiming first position 8689 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8690 // first effect claiming last position 8691 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8692 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8693 // already present 8694 8695 size_t size = mEffects.size(); 8696 size_t idx_insert = size; 8697 ssize_t idx_insert_first = -1; 8698 ssize_t idx_insert_last = -1; 8699 8700 for (size_t i = 0; i < size; i++) { 8701 effect_descriptor_t d = mEffects[i]->desc(); 8702 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8703 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8704 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8705 // check invalid effect chaining combinations 8706 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8707 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8708 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8709 return INVALID_OPERATION; 8710 } 8711 // remember position of first insert effect and by default 8712 // select this as insert position for new effect 8713 if (idx_insert == size) { 8714 idx_insert = i; 8715 } 8716 // remember position of last insert effect claiming 8717 // first position 8718 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8719 idx_insert_first = i; 8720 } 8721 // remember position of first insert effect claiming 8722 // last position 8723 if (iPref == EFFECT_FLAG_INSERT_LAST && 8724 idx_insert_last == -1) { 8725 idx_insert_last = i; 8726 } 8727 } 8728 } 8729 8730 // modify idx_insert from first position if needed 8731 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8732 if (idx_insert_last != -1) { 8733 idx_insert = idx_insert_last; 8734 } else { 8735 idx_insert = size; 8736 } 8737 } else { 8738 if (idx_insert_first != -1) { 8739 idx_insert = idx_insert_first + 1; 8740 } 8741 } 8742 8743 // always read samples from chain input buffer 8744 effect->setInBuffer(mInBuffer); 8745 8746 // if last effect in the chain, output samples to chain 8747 // output buffer, otherwise to chain input buffer 8748 if (idx_insert == size) { 8749 if (idx_insert != 0) { 8750 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8751 mEffects[idx_insert-1]->configure(); 8752 } 8753 effect->setOutBuffer(mOutBuffer); 8754 } else { 8755 effect->setOutBuffer(mInBuffer); 8756 } 8757 mEffects.insertAt(effect, idx_insert); 8758 8759 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8760 } 8761 effect->configure(); 8762 return NO_ERROR; 8763} 8764 8765// removeEffect_l() must be called with PlaybackThread::mLock held 8766size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8767{ 8768 Mutex::Autolock _l(mLock); 8769 size_t size = mEffects.size(); 8770 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8771 8772 for (size_t i = 0; i < size; i++) { 8773 if (effect == mEffects[i]) { 8774 // calling stop here will remove pre-processing effect from the audio HAL. 8775 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8776 // the middle of a read from audio HAL 8777 if (mEffects[i]->state() == EffectModule::ACTIVE || 8778 mEffects[i]->state() == EffectModule::STOPPING) { 8779 mEffects[i]->stop(); 8780 } 8781 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8782 delete[] effect->inBuffer(); 8783 } else { 8784 if (i == size - 1 && i != 0) { 8785 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8786 mEffects[i - 1]->configure(); 8787 } 8788 } 8789 mEffects.removeAt(i); 8790 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8791 break; 8792 } 8793 } 8794 8795 return mEffects.size(); 8796} 8797 8798// setDevice_l() must be called with PlaybackThread::mLock held 8799void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8800{ 8801 size_t size = mEffects.size(); 8802 for (size_t i = 0; i < size; i++) { 8803 mEffects[i]->setDevice(device); 8804 } 8805} 8806 8807// setMode_l() must be called with PlaybackThread::mLock held 8808void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8809{ 8810 size_t size = mEffects.size(); 8811 for (size_t i = 0; i < size; i++) { 8812 mEffects[i]->setMode(mode); 8813 } 8814} 8815 8816// setVolume_l() must be called with PlaybackThread::mLock held 8817bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8818{ 8819 uint32_t newLeft = *left; 8820 uint32_t newRight = *right; 8821 bool hasControl = false; 8822 int ctrlIdx = -1; 8823 size_t size = mEffects.size(); 8824 8825 // first update volume controller 8826 for (size_t i = size; i > 0; i--) { 8827 if (mEffects[i - 1]->isProcessEnabled() && 8828 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8829 ctrlIdx = i - 1; 8830 hasControl = true; 8831 break; 8832 } 8833 } 8834 8835 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8836 if (hasControl) { 8837 *left = mNewLeftVolume; 8838 *right = mNewRightVolume; 8839 } 8840 return hasControl; 8841 } 8842 8843 mVolumeCtrlIdx = ctrlIdx; 8844 mLeftVolume = newLeft; 8845 mRightVolume = newRight; 8846 8847 // second get volume update from volume controller 8848 if (ctrlIdx >= 0) { 8849 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8850 mNewLeftVolume = newLeft; 8851 mNewRightVolume = newRight; 8852 } 8853 // then indicate volume to all other effects in chain. 8854 // Pass altered volume to effects before volume controller 8855 // and requested volume to effects after controller 8856 uint32_t lVol = newLeft; 8857 uint32_t rVol = newRight; 8858 8859 for (size_t i = 0; i < size; i++) { 8860 if ((int)i == ctrlIdx) continue; 8861 // this also works for ctrlIdx == -1 when there is no volume controller 8862 if ((int)i > ctrlIdx) { 8863 lVol = *left; 8864 rVol = *right; 8865 } 8866 mEffects[i]->setVolume(&lVol, &rVol, false); 8867 } 8868 *left = newLeft; 8869 *right = newRight; 8870 8871 return hasControl; 8872} 8873 8874status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8875{ 8876 const size_t SIZE = 256; 8877 char buffer[SIZE]; 8878 String8 result; 8879 8880 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8881 result.append(buffer); 8882 8883 bool locked = tryLock(mLock); 8884 // failed to lock - AudioFlinger is probably deadlocked 8885 if (!locked) { 8886 result.append("\tCould not lock mutex:\n"); 8887 } 8888 8889 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8890 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8891 mEffects.size(), 8892 (uint32_t)mInBuffer, 8893 (uint32_t)mOutBuffer, 8894 mActiveTrackCnt); 8895 result.append(buffer); 8896 write(fd, result.string(), result.size()); 8897 8898 for (size_t i = 0; i < mEffects.size(); ++i) { 8899 sp<EffectModule> effect = mEffects[i]; 8900 if (effect != 0) { 8901 effect->dump(fd, args); 8902 } 8903 } 8904 8905 if (locked) { 8906 mLock.unlock(); 8907 } 8908 8909 return NO_ERROR; 8910} 8911 8912// must be called with ThreadBase::mLock held 8913void AudioFlinger::EffectChain::setEffectSuspended_l( 8914 const effect_uuid_t *type, bool suspend) 8915{ 8916 sp<SuspendedEffectDesc> desc; 8917 // use effect type UUID timelow as key as there is no real risk of identical 8918 // timeLow fields among effect type UUIDs. 8919 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8920 if (suspend) { 8921 if (index >= 0) { 8922 desc = mSuspendedEffects.valueAt(index); 8923 } else { 8924 desc = new SuspendedEffectDesc(); 8925 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8926 mSuspendedEffects.add(type->timeLow, desc); 8927 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8928 } 8929 if (desc->mRefCount++ == 0) { 8930 sp<EffectModule> effect = getEffectIfEnabled(type); 8931 if (effect != 0) { 8932 desc->mEffect = effect; 8933 effect->setSuspended(true); 8934 effect->setEnabled(false); 8935 } 8936 } 8937 } else { 8938 if (index < 0) { 8939 return; 8940 } 8941 desc = mSuspendedEffects.valueAt(index); 8942 if (desc->mRefCount <= 0) { 8943 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8944 desc->mRefCount = 1; 8945 } 8946 if (--desc->mRefCount == 0) { 8947 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8948 if (desc->mEffect != 0) { 8949 sp<EffectModule> effect = desc->mEffect.promote(); 8950 if (effect != 0) { 8951 effect->setSuspended(false); 8952 sp<EffectHandle> handle = effect->controlHandle(); 8953 if (handle != 0) { 8954 effect->setEnabled(handle->enabled()); 8955 } 8956 } 8957 desc->mEffect.clear(); 8958 } 8959 mSuspendedEffects.removeItemsAt(index); 8960 } 8961 } 8962} 8963 8964// must be called with ThreadBase::mLock held 8965void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8966{ 8967 sp<SuspendedEffectDesc> desc; 8968 8969 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8970 if (suspend) { 8971 if (index >= 0) { 8972 desc = mSuspendedEffects.valueAt(index); 8973 } else { 8974 desc = new SuspendedEffectDesc(); 8975 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8976 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8977 } 8978 if (desc->mRefCount++ == 0) { 8979 Vector< sp<EffectModule> > effects; 8980 getSuspendEligibleEffects(effects); 8981 for (size_t i = 0; i < effects.size(); i++) { 8982 setEffectSuspended_l(&effects[i]->desc().type, true); 8983 } 8984 } 8985 } else { 8986 if (index < 0) { 8987 return; 8988 } 8989 desc = mSuspendedEffects.valueAt(index); 8990 if (desc->mRefCount <= 0) { 8991 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8992 desc->mRefCount = 1; 8993 } 8994 if (--desc->mRefCount == 0) { 8995 Vector<const effect_uuid_t *> types; 8996 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8997 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8998 continue; 8999 } 9000 types.add(&mSuspendedEffects.valueAt(i)->mType); 9001 } 9002 for (size_t i = 0; i < types.size(); i++) { 9003 setEffectSuspended_l(types[i], false); 9004 } 9005 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9006 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9007 } 9008 } 9009} 9010 9011 9012// The volume effect is used for automated tests only 9013#ifndef OPENSL_ES_H_ 9014static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9015 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9016const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9017#endif //OPENSL_ES_H_ 9018 9019bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9020{ 9021 // auxiliary effects and visualizer are never suspended on output mix 9022 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9023 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9024 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9025 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9026 return false; 9027 } 9028 return true; 9029} 9030 9031void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9032{ 9033 effects.clear(); 9034 for (size_t i = 0; i < mEffects.size(); i++) { 9035 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9036 effects.add(mEffects[i]); 9037 } 9038 } 9039} 9040 9041sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9042 const effect_uuid_t *type) 9043{ 9044 sp<EffectModule> effect = getEffectFromType_l(type); 9045 return effect != 0 && effect->isEnabled() ? effect : 0; 9046} 9047 9048void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9049 bool enabled) 9050{ 9051 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9052 if (enabled) { 9053 if (index < 0) { 9054 // if the effect is not suspend check if all effects are suspended 9055 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9056 if (index < 0) { 9057 return; 9058 } 9059 if (!isEffectEligibleForSuspend(effect->desc())) { 9060 return; 9061 } 9062 setEffectSuspended_l(&effect->desc().type, enabled); 9063 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9064 if (index < 0) { 9065 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9066 return; 9067 } 9068 } 9069 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9070 effect->desc().type.timeLow); 9071 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9072 // if effect is requested to suspended but was not yet enabled, supend it now. 9073 if (desc->mEffect == 0) { 9074 desc->mEffect = effect; 9075 effect->setEnabled(false); 9076 effect->setSuspended(true); 9077 } 9078 } else { 9079 if (index < 0) { 9080 return; 9081 } 9082 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9083 effect->desc().type.timeLow); 9084 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9085 desc->mEffect.clear(); 9086 effect->setSuspended(false); 9087 } 9088} 9089 9090#undef LOG_TAG 9091#define LOG_TAG "AudioFlinger" 9092 9093// ---------------------------------------------------------------------------- 9094 9095status_t AudioFlinger::onTransact( 9096 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9097{ 9098 return BnAudioFlinger::onTransact(code, data, reply, flags); 9099} 9100 9101}; // namespace android 9102