AudioFlinger.cpp revision 529e888738a91ca70cbdeeabd982f8fb2947780c
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 <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#ifdef HAVE_REQUEST_PRIORITY 87#include "SchedulingPolicyService.h" 88#endif 89 90#ifdef SOAKER 91#include "Soaker.h" 92#endif 93 94// ---------------------------------------------------------------------------- 95 96// Note: the following macro is used for extremely verbose logging message. In 97// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 98// 0; but one side effect of this is to turn all LOGV's as well. Some messages 99// are so verbose that we want to suppress them even when we have ALOG_ASSERT 100// turned on. Do not uncomment the #def below unless you really know what you 101// are doing and want to see all of the extremely verbose messages. 102//#define VERY_VERY_VERBOSE_LOGGING 103#ifdef VERY_VERY_VERBOSE_LOGGING 104#define ALOGVV ALOGV 105#else 106#define ALOGVV(a...) do { } while(0) 107#endif 108 109namespace android { 110 111static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 112static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 113 114static const float MAX_GAIN = 4096.0f; 115static const uint32_t MAX_GAIN_INT = 0x1000; 116 117// retry counts for buffer fill timeout 118// 50 * ~20msecs = 1 second 119static const int8_t kMaxTrackRetries = 50; 120static const int8_t kMaxTrackStartupRetries = 50; 121// allow less retry attempts on direct output thread. 122// direct outputs can be a scarce resource in audio hardware and should 123// be released as quickly as possible. 124static const int8_t kMaxTrackRetriesDirect = 2; 125 126static const int kDumpLockRetries = 50; 127static const int kDumpLockSleepUs = 20000; 128 129// don't warn about blocked writes or record buffer overflows more often than this 130static const nsecs_t kWarningThrottleNs = seconds(5); 131 132// RecordThread loop sleep time upon application overrun or audio HAL read error 133static const int kRecordThreadSleepUs = 5000; 134 135// maximum time to wait for setParameters to complete 136static const nsecs_t kSetParametersTimeoutNs = seconds(2); 137 138// minimum sleep time for the mixer thread loop when tracks are active but in underrun 139static const uint32_t kMinThreadSleepTimeUs = 5000; 140// maximum divider applied to the active sleep time in the mixer thread loop 141static const uint32_t kMaxThreadSleepTimeShift = 2; 142 143// minimum normal mix buffer size, expressed in milliseconds rather than frames 144static const uint32_t kMinNormalMixBufferSizeMs = 20; 145// maximum normal mix buffer size 146static const uint32_t kMaxNormalMixBufferSizeMs = 24; 147 148nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 149 150// Whether to use fast mixer 151static const enum { 152 FastMixer_Never, // never initialize or use: for debugging only 153 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 154 // normal mixer multiplier is 1 155 FastMixer_Static, // initialize if needed, then use all the time if initialized, 156 // multiplier is calculated based on min & max normal mixer buffer size 157 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 158 // multiplier is calculated based on min & max normal mixer buffer size 159 // FIXME for FastMixer_Dynamic: 160 // Supporting this option will require fixing HALs that can't handle large writes. 161 // For example, one HAL implementation returns an error from a large write, 162 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 163 // We could either fix the HAL implementations, or provide a wrapper that breaks 164 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 165} kUseFastMixer = FastMixer_Static; 166 167// ---------------------------------------------------------------------------- 168 169#ifdef ADD_BATTERY_DATA 170// To collect the amplifier usage 171static void addBatteryData(uint32_t params) { 172 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 173 if (service == NULL) { 174 // it already logged 175 return; 176 } 177 178 service->addBatteryData(params); 179} 180#endif 181 182static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 183{ 184 const hw_module_t *mod; 185 int rc; 186 187 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 188 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 189 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 190 if (rc) { 191 goto out; 192 } 193 rc = audio_hw_device_open(mod, dev); 194 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 195 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 196 if (rc) { 197 goto out; 198 } 199 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 200 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 201 rc = BAD_VALUE; 202 goto out; 203 } 204 return 0; 205 206out: 207 *dev = NULL; 208 return rc; 209} 210 211// ---------------------------------------------------------------------------- 212 213AudioFlinger::AudioFlinger() 214 : BnAudioFlinger(), 215 mPrimaryHardwareDev(NULL), 216 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 217 mMasterVolume(1.0f), 218 mMasterVolumeSupportLvl(MVS_NONE), 219 mMasterMute(false), 220 mNextUniqueId(1), 221 mMode(AUDIO_MODE_INVALID), 222 mBtNrecIsOff(false) 223{ 224} 225 226void AudioFlinger::onFirstRef() 227{ 228 int rc = 0; 229 230 Mutex::Autolock _l(mLock); 231 232 /* TODO: move all this work into an Init() function */ 233 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 234 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 235 uint32_t int_val; 236 if (1 == sscanf(val_str, "%u", &int_val)) { 237 mStandbyTimeInNsecs = milliseconds(int_val); 238 ALOGI("Using %u mSec as standby time.", int_val); 239 } else { 240 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 241 ALOGI("Using default %u mSec as standby time.", 242 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 243 } 244 } 245 246 mMode = AUDIO_MODE_NORMAL; 247 mMasterVolumeSW = 1.0; 248 mMasterVolume = 1.0; 249 mHardwareStatus = AUDIO_HW_IDLE; 250} 251 252AudioFlinger::~AudioFlinger() 253{ 254 255 while (!mRecordThreads.isEmpty()) { 256 // closeInput() will remove first entry from mRecordThreads 257 closeInput(mRecordThreads.keyAt(0)); 258 } 259 while (!mPlaybackThreads.isEmpty()) { 260 // closeOutput() will remove first entry from mPlaybackThreads 261 closeOutput(mPlaybackThreads.keyAt(0)); 262 } 263 264 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 265 // no mHardwareLock needed, as there are no other references to this 266 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 267 delete mAudioHwDevs.valueAt(i); 268 } 269} 270 271static const char * const audio_interfaces[] = { 272 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 273 AUDIO_HARDWARE_MODULE_ID_A2DP, 274 AUDIO_HARDWARE_MODULE_ID_USB, 275}; 276#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 277 278audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 279{ 280 // if module is 0, the request comes from an old policy manager and we should load 281 // well known modules 282 if (module == 0) { 283 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 284 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 285 loadHwModule_l(audio_interfaces[i]); 286 } 287 } else { 288 // check a match for the requested module handle 289 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 290 if (audioHwdevice != NULL) { 291 return audioHwdevice->hwDevice(); 292 } 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 297 if ((dev->get_supported_devices(dev) & devices) == devices) 298 return dev; 299 } 300 301 return NULL; 302} 303 304status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Global session refs:\n"); 320 result.append(" session pid count\n"); 321 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 322 AudioSessionRef *r = mAudioSessionRefs[i]; 323 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 324 result.append(buffer); 325 } 326 write(fd, result.string(), result.size()); 327 return NO_ERROR; 328} 329 330 331status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 332{ 333 const size_t SIZE = 256; 334 char buffer[SIZE]; 335 String8 result; 336 hardware_call_state hardwareStatus = mHardwareStatus; 337 338 snprintf(buffer, SIZE, "Hardware status: %d\n" 339 "Standby Time mSec: %u\n", 340 hardwareStatus, 341 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 342 result.append(buffer); 343 write(fd, result.string(), result.size()); 344 return NO_ERROR; 345} 346 347status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 348{ 349 const size_t SIZE = 256; 350 char buffer[SIZE]; 351 String8 result; 352 snprintf(buffer, SIZE, "Permission Denial: " 353 "can't dump AudioFlinger from pid=%d, uid=%d\n", 354 IPCThreadState::self()->getCallingPid(), 355 IPCThreadState::self()->getCallingUid()); 356 result.append(buffer); 357 write(fd, result.string(), result.size()); 358 return NO_ERROR; 359} 360 361static bool tryLock(Mutex& mutex) 362{ 363 bool locked = false; 364 for (int i = 0; i < kDumpLockRetries; ++i) { 365 if (mutex.tryLock() == NO_ERROR) { 366 locked = true; 367 break; 368 } 369 usleep(kDumpLockSleepUs); 370 } 371 return locked; 372} 373 374status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 375{ 376 if (!dumpAllowed()) { 377 dumpPermissionDenial(fd, args); 378 } else { 379 // get state of hardware lock 380 bool hardwareLocked = tryLock(mHardwareLock); 381 if (!hardwareLocked) { 382 String8 result(kHardwareLockedString); 383 write(fd, result.string(), result.size()); 384 } else { 385 mHardwareLock.unlock(); 386 } 387 388 bool locked = tryLock(mLock); 389 390 // failed to lock - AudioFlinger is probably deadlocked 391 if (!locked) { 392 String8 result(kDeadlockedString); 393 write(fd, result.string(), result.size()); 394 } 395 396 dumpClients(fd, args); 397 dumpInternals(fd, args); 398 399 // dump playback threads 400 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 401 mPlaybackThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump record threads 405 for (size_t i = 0; i < mRecordThreads.size(); i++) { 406 mRecordThreads.valueAt(i)->dump(fd, args); 407 } 408 409 // dump all hardware devs 410 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 411 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 412 dev->dump(dev, fd); 413 } 414 if (locked) mLock.unlock(); 415 } 416 return NO_ERROR; 417} 418 419sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 420{ 421 // If pid is already in the mClients wp<> map, then use that entry 422 // (for which promote() is always != 0), otherwise create a new entry and Client. 423 sp<Client> client = mClients.valueFor(pid).promote(); 424 if (client == 0) { 425 client = new Client(this, pid); 426 mClients.add(pid, client); 427 } 428 429 return client; 430} 431 432// IAudioFlinger interface 433 434 435sp<IAudioTrack> AudioFlinger::createTrack( 436 pid_t pid, 437 audio_stream_type_t streamType, 438 uint32_t sampleRate, 439 audio_format_t format, 440 uint32_t channelMask, 441 int frameCount, 442 IAudioFlinger::track_flags_t flags, 443 const sp<IMemory>& sharedBuffer, 444 audio_io_handle_t output, 445 pid_t tid, 446 int *sessionId, 447 status_t *status) 448{ 449 sp<PlaybackThread::Track> track; 450 sp<TrackHandle> trackHandle; 451 sp<Client> client; 452 status_t lStatus; 453 int lSessionId; 454 455 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 456 // but if someone uses binder directly they could bypass that and cause us to crash 457 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 458 ALOGE("createTrack() invalid stream type %d", streamType); 459 lStatus = BAD_VALUE; 460 goto Exit; 461 } 462 463 { 464 Mutex::Autolock _l(mLock); 465 PlaybackThread *thread = checkPlaybackThread_l(output); 466 PlaybackThread *effectThread = NULL; 467 if (thread == NULL) { 468 ALOGE("unknown output thread"); 469 lStatus = BAD_VALUE; 470 goto Exit; 471 } 472 473 client = registerPid_l(pid); 474 475 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 476 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 477 // check if an effect chain with the same session ID is present on another 478 // output thread and move it here. 479 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 480 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 481 if (mPlaybackThreads.keyAt(i) != output) { 482 uint32_t sessions = t->hasAudioSession(*sessionId); 483 if (sessions & PlaybackThread::EFFECT_SESSION) { 484 effectThread = t.get(); 485 break; 486 } 487 } 488 } 489 lSessionId = *sessionId; 490 } else { 491 // if no audio session id is provided, create one here 492 lSessionId = nextUniqueId(); 493 if (sessionId != NULL) { 494 *sessionId = lSessionId; 495 } 496 } 497 ALOGV("createTrack() lSessionId: %d", lSessionId); 498 499 track = thread->createTrack_l(client, streamType, sampleRate, format, 500 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 501 502 // move effect chain to this output thread if an effect on same session was waiting 503 // for a track to be created 504 if (lStatus == NO_ERROR && effectThread != NULL) { 505 Mutex::Autolock _dl(thread->mLock); 506 Mutex::Autolock _sl(effectThread->mLock); 507 moveEffectChain_l(lSessionId, effectThread, thread, true); 508 } 509 510 // Look for sync events awaiting for a session to be used. 511 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 512 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 513 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 514 if (lStatus == NO_ERROR) { 515 track->setSyncEvent(mPendingSyncEvents[i]); 516 } else { 517 mPendingSyncEvents[i]->cancel(); 518 } 519 mPendingSyncEvents.removeAt(i); 520 i--; 521 } 522 } 523 } 524 } 525 if (lStatus == NO_ERROR) { 526 trackHandle = new TrackHandle(track); 527 } else { 528 // remove local strong reference to Client before deleting the Track so that the Client 529 // destructor is called by the TrackBase destructor with mLock held 530 client.clear(); 531 track.clear(); 532 } 533 534Exit: 535 if (status != NULL) { 536 *status = lStatus; 537 } 538 return trackHandle; 539} 540 541uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 542{ 543 Mutex::Autolock _l(mLock); 544 PlaybackThread *thread = checkPlaybackThread_l(output); 545 if (thread == NULL) { 546 ALOGW("sampleRate() unknown thread %d", output); 547 return 0; 548 } 549 return thread->sampleRate(); 550} 551 552int AudioFlinger::channelCount(audio_io_handle_t output) const 553{ 554 Mutex::Autolock _l(mLock); 555 PlaybackThread *thread = checkPlaybackThread_l(output); 556 if (thread == NULL) { 557 ALOGW("channelCount() unknown thread %d", output); 558 return 0; 559 } 560 return thread->channelCount(); 561} 562 563audio_format_t AudioFlinger::format(audio_io_handle_t output) const 564{ 565 Mutex::Autolock _l(mLock); 566 PlaybackThread *thread = checkPlaybackThread_l(output); 567 if (thread == NULL) { 568 ALOGW("format() unknown thread %d", output); 569 return AUDIO_FORMAT_INVALID; 570 } 571 return thread->format(); 572} 573 574size_t AudioFlinger::frameCount(audio_io_handle_t output) const 575{ 576 Mutex::Autolock _l(mLock); 577 PlaybackThread *thread = checkPlaybackThread_l(output); 578 if (thread == NULL) { 579 ALOGW("frameCount() unknown thread %d", output); 580 return 0; 581 } 582 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 583 // should examine all callers and fix them to handle smaller counts 584 return thread->frameCount(); 585} 586 587uint32_t AudioFlinger::latency(audio_io_handle_t output) const 588{ 589 Mutex::Autolock _l(mLock); 590 PlaybackThread *thread = checkPlaybackThread_l(output); 591 if (thread == NULL) { 592 ALOGW("latency() unknown thread %d", output); 593 return 0; 594 } 595 return thread->latency(); 596} 597 598status_t AudioFlinger::setMasterVolume(float value) 599{ 600 status_t ret = initCheck(); 601 if (ret != NO_ERROR) { 602 return ret; 603 } 604 605 // check calling permissions 606 if (!settingsAllowed()) { 607 return PERMISSION_DENIED; 608 } 609 610 float swmv = value; 611 612 Mutex::Autolock _l(mLock); 613 614 // when hw supports master volume, don't scale in sw mixer 615 if (MVS_NONE != mMasterVolumeSupportLvl) { 616 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 617 AutoMutex lock(mHardwareLock); 618 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 619 620 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 621 if (NULL != dev->set_master_volume) { 622 dev->set_master_volume(dev, value); 623 } 624 mHardwareStatus = AUDIO_HW_IDLE; 625 } 626 627 swmv = 1.0; 628 } 629 630 mMasterVolume = value; 631 mMasterVolumeSW = swmv; 632 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 633 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 634 635 return NO_ERROR; 636} 637 638status_t AudioFlinger::setMode(audio_mode_t mode) 639{ 640 status_t ret = initCheck(); 641 if (ret != NO_ERROR) { 642 return ret; 643 } 644 645 // check calling permissions 646 if (!settingsAllowed()) { 647 return PERMISSION_DENIED; 648 } 649 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 650 ALOGW("Illegal value: setMode(%d)", mode); 651 return BAD_VALUE; 652 } 653 654 { // scope for the lock 655 AutoMutex lock(mHardwareLock); 656 mHardwareStatus = AUDIO_HW_SET_MODE; 657 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 658 mHardwareStatus = AUDIO_HW_IDLE; 659 } 660 661 if (NO_ERROR == ret) { 662 Mutex::Autolock _l(mLock); 663 mMode = mode; 664 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 665 mPlaybackThreads.valueAt(i)->setMode(mode); 666 } 667 668 return ret; 669} 670 671status_t AudioFlinger::setMicMute(bool state) 672{ 673 status_t ret = initCheck(); 674 if (ret != NO_ERROR) { 675 return ret; 676 } 677 678 // check calling permissions 679 if (!settingsAllowed()) { 680 return PERMISSION_DENIED; 681 } 682 683 AutoMutex lock(mHardwareLock); 684 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 685 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 686 mHardwareStatus = AUDIO_HW_IDLE; 687 return ret; 688} 689 690bool AudioFlinger::getMicMute() const 691{ 692 status_t ret = initCheck(); 693 if (ret != NO_ERROR) { 694 return false; 695 } 696 697 bool state = AUDIO_MODE_INVALID; 698 AutoMutex lock(mHardwareLock); 699 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 700 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 701 mHardwareStatus = AUDIO_HW_IDLE; 702 return state; 703} 704 705status_t AudioFlinger::setMasterMute(bool muted) 706{ 707 // check calling permissions 708 if (!settingsAllowed()) { 709 return PERMISSION_DENIED; 710 } 711 712 Mutex::Autolock _l(mLock); 713 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 714 mMasterMute = muted; 715 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 716 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 717 718 return NO_ERROR; 719} 720 721float AudioFlinger::masterVolume() const 722{ 723 Mutex::Autolock _l(mLock); 724 return masterVolume_l(); 725} 726 727float AudioFlinger::masterVolumeSW() const 728{ 729 Mutex::Autolock _l(mLock); 730 return masterVolumeSW_l(); 731} 732 733bool AudioFlinger::masterMute() const 734{ 735 Mutex::Autolock _l(mLock); 736 return masterMute_l(); 737} 738 739float AudioFlinger::masterVolume_l() const 740{ 741 if (MVS_FULL == mMasterVolumeSupportLvl) { 742 float ret_val; 743 AutoMutex lock(mHardwareLock); 744 745 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 746 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 747 (NULL != mPrimaryHardwareDev->get_master_volume), 748 "can't get master volume"); 749 750 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 751 mHardwareStatus = AUDIO_HW_IDLE; 752 return ret_val; 753 } 754 755 return mMasterVolume; 756} 757 758status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 759 audio_io_handle_t output) 760{ 761 // check calling permissions 762 if (!settingsAllowed()) { 763 return PERMISSION_DENIED; 764 } 765 766 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 767 ALOGE("setStreamVolume() invalid stream %d", stream); 768 return BAD_VALUE; 769 } 770 771 AutoMutex lock(mLock); 772 PlaybackThread *thread = NULL; 773 if (output) { 774 thread = checkPlaybackThread_l(output); 775 if (thread == NULL) { 776 return BAD_VALUE; 777 } 778 } 779 780 mStreamTypes[stream].volume = value; 781 782 if (thread == NULL) { 783 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 784 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 785 } 786 } else { 787 thread->setStreamVolume(stream, value); 788 } 789 790 return NO_ERROR; 791} 792 793status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 794{ 795 // check calling permissions 796 if (!settingsAllowed()) { 797 return PERMISSION_DENIED; 798 } 799 800 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 801 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 802 ALOGE("setStreamMute() invalid stream %d", stream); 803 return BAD_VALUE; 804 } 805 806 AutoMutex lock(mLock); 807 mStreamTypes[stream].mute = muted; 808 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 809 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 810 811 return NO_ERROR; 812} 813 814float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return 0.0f; 818 } 819 820 AutoMutex lock(mLock); 821 float volume; 822 if (output) { 823 PlaybackThread *thread = checkPlaybackThread_l(output); 824 if (thread == NULL) { 825 return 0.0f; 826 } 827 volume = thread->streamVolume(stream); 828 } else { 829 volume = streamVolume_l(stream); 830 } 831 832 return volume; 833} 834 835bool AudioFlinger::streamMute(audio_stream_type_t stream) const 836{ 837 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 838 return true; 839 } 840 841 AutoMutex lock(mLock); 842 return streamMute_l(stream); 843} 844 845status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 846{ 847 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 848 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 849 // check calling permissions 850 if (!settingsAllowed()) { 851 return PERMISSION_DENIED; 852 } 853 854 // ioHandle == 0 means the parameters are global to the audio hardware interface 855 if (ioHandle == 0) { 856 Mutex::Autolock _l(mLock); 857 status_t final_result = NO_ERROR; 858 { 859 AutoMutex lock(mHardwareLock); 860 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 861 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 862 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 863 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 864 final_result = result ?: final_result; 865 } 866 mHardwareStatus = AUDIO_HW_IDLE; 867 } 868 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 869 AudioParameter param = AudioParameter(keyValuePairs); 870 String8 value; 871 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 872 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 873 if (mBtNrecIsOff != btNrecIsOff) { 874 for (size_t i = 0; i < mRecordThreads.size(); i++) { 875 sp<RecordThread> thread = mRecordThreads.valueAt(i); 876 RecordThread::RecordTrack *track = thread->track(); 877 if (track != NULL) { 878 audio_devices_t device = (audio_devices_t)( 879 thread->device() & AUDIO_DEVICE_IN_ALL); 880 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 881 thread->setEffectSuspended(FX_IID_AEC, 882 suspend, 883 track->sessionId()); 884 thread->setEffectSuspended(FX_IID_NS, 885 suspend, 886 track->sessionId()); 887 } 888 } 889 mBtNrecIsOff = btNrecIsOff; 890 } 891 } 892 return final_result; 893 } 894 895 // hold a strong ref on thread in case closeOutput() or closeInput() is called 896 // and the thread is exited once the lock is released 897 sp<ThreadBase> thread; 898 { 899 Mutex::Autolock _l(mLock); 900 thread = checkPlaybackThread_l(ioHandle); 901 if (thread == NULL) { 902 thread = checkRecordThread_l(ioHandle); 903 } else if (thread == primaryPlaybackThread_l()) { 904 // indicate output device change to all input threads for pre processing 905 AudioParameter param = AudioParameter(keyValuePairs); 906 int value; 907 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 908 (value != 0)) { 909 for (size_t i = 0; i < mRecordThreads.size(); i++) { 910 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 911 } 912 } 913 } 914 } 915 if (thread != 0) { 916 return thread->setParameters(keyValuePairs); 917 } 918 return BAD_VALUE; 919} 920 921String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 922{ 923// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 924// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 925 926 Mutex::Autolock _l(mLock); 927 928 if (ioHandle == 0) { 929 String8 out_s8; 930 931 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 932 char *s; 933 { 934 AutoMutex lock(mHardwareLock); 935 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 936 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 937 s = dev->get_parameters(dev, keys.string()); 938 mHardwareStatus = AUDIO_HW_IDLE; 939 } 940 out_s8 += String8(s ? s : ""); 941 free(s); 942 } 943 return out_s8; 944 } 945 946 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 947 if (playbackThread != NULL) { 948 return playbackThread->getParameters(keys); 949 } 950 RecordThread *recordThread = checkRecordThread_l(ioHandle); 951 if (recordThread != NULL) { 952 return recordThread->getParameters(keys); 953 } 954 return String8(""); 955} 956 957size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 958{ 959 status_t ret = initCheck(); 960 if (ret != NO_ERROR) { 961 return 0; 962 } 963 964 AutoMutex lock(mHardwareLock); 965 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 966 struct audio_config config = { 967 sample_rate: sampleRate, 968 channel_mask: audio_channel_in_mask_from_count(channelCount), 969 format: format, 970 }; 971 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 972 mHardwareStatus = AUDIO_HW_IDLE; 973 return size; 974} 975 976unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 977{ 978 if (ioHandle == 0) { 979 return 0; 980 } 981 982 Mutex::Autolock _l(mLock); 983 984 RecordThread *recordThread = checkRecordThread_l(ioHandle); 985 if (recordThread != NULL) { 986 return recordThread->getInputFramesLost(); 987 } 988 return 0; 989} 990 991status_t AudioFlinger::setVoiceVolume(float value) 992{ 993 status_t ret = initCheck(); 994 if (ret != NO_ERROR) { 995 return ret; 996 } 997 998 // check calling permissions 999 if (!settingsAllowed()) { 1000 return PERMISSION_DENIED; 1001 } 1002 1003 AutoMutex lock(mHardwareLock); 1004 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1005 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1006 mHardwareStatus = AUDIO_HW_IDLE; 1007 1008 return ret; 1009} 1010 1011status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1012 audio_io_handle_t output) const 1013{ 1014 status_t status; 1015 1016 Mutex::Autolock _l(mLock); 1017 1018 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1019 if (playbackThread != NULL) { 1020 return playbackThread->getRenderPosition(halFrames, dspFrames); 1021 } 1022 1023 return BAD_VALUE; 1024} 1025 1026void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1027{ 1028 1029 Mutex::Autolock _l(mLock); 1030 1031 pid_t pid = IPCThreadState::self()->getCallingPid(); 1032 if (mNotificationClients.indexOfKey(pid) < 0) { 1033 sp<NotificationClient> notificationClient = new NotificationClient(this, 1034 client, 1035 pid); 1036 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1037 1038 mNotificationClients.add(pid, notificationClient); 1039 1040 sp<IBinder> binder = client->asBinder(); 1041 binder->linkToDeath(notificationClient); 1042 1043 // the config change is always sent from playback or record threads to avoid deadlock 1044 // with AudioSystem::gLock 1045 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1046 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1047 } 1048 1049 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1050 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1051 } 1052 } 1053} 1054 1055void AudioFlinger::removeNotificationClient(pid_t pid) 1056{ 1057 Mutex::Autolock _l(mLock); 1058 1059 mNotificationClients.removeItem(pid); 1060 1061 ALOGV("%d died, releasing its sessions", pid); 1062 size_t num = mAudioSessionRefs.size(); 1063 bool removed = false; 1064 for (size_t i = 0; i< num; ) { 1065 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1066 ALOGV(" pid %d @ %d", ref->mPid, i); 1067 if (ref->mPid == pid) { 1068 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1069 mAudioSessionRefs.removeAt(i); 1070 delete ref; 1071 removed = true; 1072 num--; 1073 } else { 1074 i++; 1075 } 1076 } 1077 if (removed) { 1078 purgeStaleEffects_l(); 1079 } 1080} 1081 1082// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1083void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1084{ 1085 size_t size = mNotificationClients.size(); 1086 for (size_t i = 0; i < size; i++) { 1087 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1088 param2); 1089 } 1090} 1091 1092// removeClient_l() must be called with AudioFlinger::mLock held 1093void AudioFlinger::removeClient_l(pid_t pid) 1094{ 1095 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1096 mClients.removeItem(pid); 1097} 1098 1099 1100// ---------------------------------------------------------------------------- 1101 1102AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1103 uint32_t device, type_t type) 1104 : Thread(false), 1105 mType(type), 1106 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1107 // mChannelMask 1108 mChannelCount(0), 1109 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1110 mParamStatus(NO_ERROR), 1111 mStandby(false), mId(id), 1112 mDevice(device), 1113 mDeathRecipient(new PMDeathRecipient(this)) 1114{ 1115} 1116 1117AudioFlinger::ThreadBase::~ThreadBase() 1118{ 1119 mParamCond.broadcast(); 1120 // do not lock the mutex in destructor 1121 releaseWakeLock_l(); 1122 if (mPowerManager != 0) { 1123 sp<IBinder> binder = mPowerManager->asBinder(); 1124 binder->unlinkToDeath(mDeathRecipient); 1125 } 1126} 1127 1128void AudioFlinger::ThreadBase::exit() 1129{ 1130 ALOGV("ThreadBase::exit"); 1131 { 1132 // This lock prevents the following race in thread (uniprocessor for illustration): 1133 // if (!exitPending()) { 1134 // // context switch from here to exit() 1135 // // exit() calls requestExit(), what exitPending() observes 1136 // // exit() calls signal(), which is dropped since no waiters 1137 // // context switch back from exit() to here 1138 // mWaitWorkCV.wait(...); 1139 // // now thread is hung 1140 // } 1141 AutoMutex lock(mLock); 1142 requestExit(); 1143 mWaitWorkCV.signal(); 1144 } 1145 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1146 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1147 requestExitAndWait(); 1148} 1149 1150status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1151{ 1152 status_t status; 1153 1154 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1155 Mutex::Autolock _l(mLock); 1156 1157 mNewParameters.add(keyValuePairs); 1158 mWaitWorkCV.signal(); 1159 // wait condition with timeout in case the thread loop has exited 1160 // before the request could be processed 1161 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1162 status = mParamStatus; 1163 mWaitWorkCV.signal(); 1164 } else { 1165 status = TIMED_OUT; 1166 } 1167 return status; 1168} 1169 1170void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1171{ 1172 Mutex::Autolock _l(mLock); 1173 sendConfigEvent_l(event, param); 1174} 1175 1176// sendConfigEvent_l() must be called with ThreadBase::mLock held 1177void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1178{ 1179 ConfigEvent configEvent; 1180 configEvent.mEvent = event; 1181 configEvent.mParam = param; 1182 mConfigEvents.add(configEvent); 1183 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1184 mWaitWorkCV.signal(); 1185} 1186 1187void AudioFlinger::ThreadBase::processConfigEvents() 1188{ 1189 mLock.lock(); 1190 while (!mConfigEvents.isEmpty()) { 1191 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1192 ConfigEvent configEvent = mConfigEvents[0]; 1193 mConfigEvents.removeAt(0); 1194 // release mLock before locking AudioFlinger mLock: lock order is always 1195 // AudioFlinger then ThreadBase to avoid cross deadlock 1196 mLock.unlock(); 1197 mAudioFlinger->mLock.lock(); 1198 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1199 mAudioFlinger->mLock.unlock(); 1200 mLock.lock(); 1201 } 1202 mLock.unlock(); 1203} 1204 1205status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1206{ 1207 const size_t SIZE = 256; 1208 char buffer[SIZE]; 1209 String8 result; 1210 1211 bool locked = tryLock(mLock); 1212 if (!locked) { 1213 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1214 write(fd, buffer, strlen(buffer)); 1215 } 1216 1217 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1218 result.append(buffer); 1219 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1220 result.append(buffer); 1221 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1222 result.append(buffer); 1223 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1224 result.append(buffer); 1225 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1226 result.append(buffer); 1227 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1228 result.append(buffer); 1229 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1236 result.append(buffer); 1237 1238 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1239 result.append(buffer); 1240 result.append(" Index Command"); 1241 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1242 snprintf(buffer, SIZE, "\n %02d ", i); 1243 result.append(buffer); 1244 result.append(mNewParameters[i]); 1245 } 1246 1247 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1248 result.append(buffer); 1249 snprintf(buffer, SIZE, " Index event param\n"); 1250 result.append(buffer); 1251 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1252 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1253 result.append(buffer); 1254 } 1255 result.append("\n"); 1256 1257 write(fd, result.string(), result.size()); 1258 1259 if (locked) { 1260 mLock.unlock(); 1261 } 1262 return NO_ERROR; 1263} 1264 1265status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1266{ 1267 const size_t SIZE = 256; 1268 char buffer[SIZE]; 1269 String8 result; 1270 1271 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1272 write(fd, buffer, strlen(buffer)); 1273 1274 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1275 sp<EffectChain> chain = mEffectChains[i]; 1276 if (chain != 0) { 1277 chain->dump(fd, args); 1278 } 1279 } 1280 return NO_ERROR; 1281} 1282 1283void AudioFlinger::ThreadBase::acquireWakeLock() 1284{ 1285 Mutex::Autolock _l(mLock); 1286 acquireWakeLock_l(); 1287} 1288 1289void AudioFlinger::ThreadBase::acquireWakeLock_l() 1290{ 1291 if (mPowerManager == 0) { 1292 // use checkService() to avoid blocking if power service is not up yet 1293 sp<IBinder> binder = 1294 defaultServiceManager()->checkService(String16("power")); 1295 if (binder == 0) { 1296 ALOGW("Thread %s cannot connect to the power manager service", mName); 1297 } else { 1298 mPowerManager = interface_cast<IPowerManager>(binder); 1299 binder->linkToDeath(mDeathRecipient); 1300 } 1301 } 1302 if (mPowerManager != 0) { 1303 sp<IBinder> binder = new BBinder(); 1304 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1305 binder, 1306 String16(mName)); 1307 if (status == NO_ERROR) { 1308 mWakeLockToken = binder; 1309 } 1310 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1311 } 1312} 1313 1314void AudioFlinger::ThreadBase::releaseWakeLock() 1315{ 1316 Mutex::Autolock _l(mLock); 1317 releaseWakeLock_l(); 1318} 1319 1320void AudioFlinger::ThreadBase::releaseWakeLock_l() 1321{ 1322 if (mWakeLockToken != 0) { 1323 ALOGV("releaseWakeLock_l() %s", mName); 1324 if (mPowerManager != 0) { 1325 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1326 } 1327 mWakeLockToken.clear(); 1328 } 1329} 1330 1331void AudioFlinger::ThreadBase::clearPowerManager() 1332{ 1333 Mutex::Autolock _l(mLock); 1334 releaseWakeLock_l(); 1335 mPowerManager.clear(); 1336} 1337 1338void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1339{ 1340 sp<ThreadBase> thread = mThread.promote(); 1341 if (thread != 0) { 1342 thread->clearPowerManager(); 1343 } 1344 ALOGW("power manager service died !!!"); 1345} 1346 1347void AudioFlinger::ThreadBase::setEffectSuspended( 1348 const effect_uuid_t *type, bool suspend, int sessionId) 1349{ 1350 Mutex::Autolock _l(mLock); 1351 setEffectSuspended_l(type, suspend, sessionId); 1352} 1353 1354void AudioFlinger::ThreadBase::setEffectSuspended_l( 1355 const effect_uuid_t *type, bool suspend, int sessionId) 1356{ 1357 sp<EffectChain> chain = getEffectChain_l(sessionId); 1358 if (chain != 0) { 1359 if (type != NULL) { 1360 chain->setEffectSuspended_l(type, suspend); 1361 } else { 1362 chain->setEffectSuspendedAll_l(suspend); 1363 } 1364 } 1365 1366 updateSuspendedSessions_l(type, suspend, sessionId); 1367} 1368 1369void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1370{ 1371 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1372 if (index < 0) { 1373 return; 1374 } 1375 1376 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1377 mSuspendedSessions.editValueAt(index); 1378 1379 for (size_t i = 0; i < sessionEffects.size(); i++) { 1380 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1381 for (int j = 0; j < desc->mRefCount; j++) { 1382 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1383 chain->setEffectSuspendedAll_l(true); 1384 } else { 1385 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1386 desc->mType.timeLow); 1387 chain->setEffectSuspended_l(&desc->mType, true); 1388 } 1389 } 1390 } 1391} 1392 1393void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1394 bool suspend, 1395 int sessionId) 1396{ 1397 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1398 1399 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1400 1401 if (suspend) { 1402 if (index >= 0) { 1403 sessionEffects = mSuspendedSessions.editValueAt(index); 1404 } else { 1405 mSuspendedSessions.add(sessionId, sessionEffects); 1406 } 1407 } else { 1408 if (index < 0) { 1409 return; 1410 } 1411 sessionEffects = mSuspendedSessions.editValueAt(index); 1412 } 1413 1414 1415 int key = EffectChain::kKeyForSuspendAll; 1416 if (type != NULL) { 1417 key = type->timeLow; 1418 } 1419 index = sessionEffects.indexOfKey(key); 1420 1421 sp<SuspendedSessionDesc> desc; 1422 if (suspend) { 1423 if (index >= 0) { 1424 desc = sessionEffects.valueAt(index); 1425 } else { 1426 desc = new SuspendedSessionDesc(); 1427 if (type != NULL) { 1428 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1429 } 1430 sessionEffects.add(key, desc); 1431 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1432 } 1433 desc->mRefCount++; 1434 } else { 1435 if (index < 0) { 1436 return; 1437 } 1438 desc = sessionEffects.valueAt(index); 1439 if (--desc->mRefCount == 0) { 1440 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1441 sessionEffects.removeItemsAt(index); 1442 if (sessionEffects.isEmpty()) { 1443 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1444 sessionId); 1445 mSuspendedSessions.removeItem(sessionId); 1446 } 1447 } 1448 } 1449 if (!sessionEffects.isEmpty()) { 1450 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1451 } 1452} 1453 1454void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1455 bool enabled, 1456 int sessionId) 1457{ 1458 Mutex::Autolock _l(mLock); 1459 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1460} 1461 1462void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1463 bool enabled, 1464 int sessionId) 1465{ 1466 if (mType != RECORD) { 1467 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1468 // another session. This gives the priority to well behaved effect control panels 1469 // and applications not using global effects. 1470 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1471 // global effects 1472 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1473 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1474 } 1475 } 1476 1477 sp<EffectChain> chain = getEffectChain_l(sessionId); 1478 if (chain != 0) { 1479 chain->checkSuspendOnEffectEnabled(effect, enabled); 1480 } 1481} 1482 1483// ---------------------------------------------------------------------------- 1484 1485AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1486 AudioStreamOut* output, 1487 audio_io_handle_t id, 1488 uint32_t device, 1489 type_t type) 1490 : ThreadBase(audioFlinger, id, device, type), 1491 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1492 // Assumes constructor is called by AudioFlinger with it's mLock held, 1493 // but it would be safer to explicitly pass initial masterMute as parameter 1494 mMasterMute(audioFlinger->masterMute_l()), 1495 // mStreamTypes[] initialized in constructor body 1496 mOutput(output), 1497 // Assumes constructor is called by AudioFlinger with it's mLock held, 1498 // but it would be safer to explicitly pass initial masterVolume as parameter 1499 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1500 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1501 mMixerStatus(MIXER_IDLE), 1502 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1503 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1504 // index 0 is reserved for normal mixer's submix 1505 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1506{ 1507 snprintf(mName, kNameLength, "AudioOut_%X", id); 1508 1509 readOutputParameters(); 1510 1511 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1512 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1513 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1514 stream = (audio_stream_type_t) (stream + 1)) { 1515 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1516 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1517 } 1518 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1519 // because mAudioFlinger doesn't have one to copy from 1520} 1521 1522AudioFlinger::PlaybackThread::~PlaybackThread() 1523{ 1524 delete [] mMixBuffer; 1525} 1526 1527status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1528{ 1529 dumpInternals(fd, args); 1530 dumpTracks(fd, args); 1531 dumpEffectChains(fd, args); 1532 return NO_ERROR; 1533} 1534 1535status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1536{ 1537 const size_t SIZE = 256; 1538 char buffer[SIZE]; 1539 String8 result; 1540 1541 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1542 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1543 const stream_type_t *st = &mStreamTypes[i]; 1544 if (i > 0) { 1545 result.appendFormat(", "); 1546 } 1547 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1548 if (st->mute) { 1549 result.append("M"); 1550 } 1551 } 1552 result.append("\n"); 1553 write(fd, result.string(), result.length()); 1554 result.clear(); 1555 1556 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1557 result.append(buffer); 1558 Track::appendDumpHeader(result); 1559 for (size_t i = 0; i < mTracks.size(); ++i) { 1560 sp<Track> track = mTracks[i]; 1561 if (track != 0) { 1562 track->dump(buffer, SIZE); 1563 result.append(buffer); 1564 } 1565 } 1566 1567 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1568 result.append(buffer); 1569 Track::appendDumpHeader(result); 1570 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1571 sp<Track> track = mActiveTracks[i].promote(); 1572 if (track != 0) { 1573 track->dump(buffer, SIZE); 1574 result.append(buffer); 1575 } 1576 } 1577 write(fd, result.string(), result.size()); 1578 1579 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1580 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1581 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1582 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1583 1584 return NO_ERROR; 1585} 1586 1587status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1588{ 1589 const size_t SIZE = 256; 1590 char buffer[SIZE]; 1591 String8 result; 1592 1593 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1594 result.append(buffer); 1595 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1596 result.append(buffer); 1597 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1598 result.append(buffer); 1599 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1600 result.append(buffer); 1601 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1602 result.append(buffer); 1603 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1604 result.append(buffer); 1605 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1606 result.append(buffer); 1607 write(fd, result.string(), result.size()); 1608 1609 dumpBase(fd, args); 1610 1611 return NO_ERROR; 1612} 1613 1614// Thread virtuals 1615status_t AudioFlinger::PlaybackThread::readyToRun() 1616{ 1617 status_t status = initCheck(); 1618 if (status == NO_ERROR) { 1619 ALOGI("AudioFlinger's thread %p ready to run", this); 1620 } else { 1621 ALOGE("No working audio driver found."); 1622 } 1623 return status; 1624} 1625 1626void AudioFlinger::PlaybackThread::onFirstRef() 1627{ 1628 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1629} 1630 1631// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1632sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1633 const sp<AudioFlinger::Client>& client, 1634 audio_stream_type_t streamType, 1635 uint32_t sampleRate, 1636 audio_format_t format, 1637 uint32_t channelMask, 1638 int frameCount, 1639 const sp<IMemory>& sharedBuffer, 1640 int sessionId, 1641 IAudioFlinger::track_flags_t flags, 1642 pid_t tid, 1643 status_t *status) 1644{ 1645 sp<Track> track; 1646 status_t lStatus; 1647 1648 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1649 1650 // client expresses a preference for FAST, but we get the final say 1651 if (flags & IAudioFlinger::TRACK_FAST) { 1652 if ( 1653 // not timed 1654 (!isTimed) && 1655 // either of these use cases: 1656 ( 1657 // use case 1: shared buffer with any frame count 1658 ( 1659 (sharedBuffer != 0) 1660 ) || 1661 // use case 2: callback handler and frame count is default or at least as large as HAL 1662 ( 1663 (tid != -1) && 1664 ((frameCount == 0) || 1665 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1666 ) 1667 ) && 1668 // PCM data 1669 audio_is_linear_pcm(format) && 1670 // mono or stereo 1671 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1672 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1673#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1674 // hardware sample rate 1675 (sampleRate == mSampleRate) && 1676#endif 1677 // normal mixer has an associated fast mixer 1678 hasFastMixer() && 1679 // there are sufficient fast track slots available 1680 (mFastTrackAvailMask != 0) 1681 // FIXME test that MixerThread for this fast track has a capable output HAL 1682 // FIXME add a permission test also? 1683 ) { 1684 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1685 if (frameCount == 0) { 1686 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1687 } 1688 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1689 frameCount, mFrameCount); 1690 } else { 1691 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1692 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1693 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1694 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1695 audio_is_linear_pcm(format), 1696 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1697 flags &= ~IAudioFlinger::TRACK_FAST; 1698 // For compatibility with AudioTrack calculation, buffer depth is forced 1699 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1700 // This is probably too conservative, but legacy application code may depend on it. 1701 // If you change this calculation, also review the start threshold which is related. 1702 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1703 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1704 if (minBufCount < 2) { 1705 minBufCount = 2; 1706 } 1707 int minFrameCount = mNormalFrameCount * minBufCount; 1708 if (frameCount < minFrameCount) { 1709 frameCount = minFrameCount; 1710 } 1711 } 1712 } 1713 1714 if (mType == DIRECT) { 1715 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1716 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1717 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1718 "for output %p with format %d", 1719 sampleRate, format, channelMask, mOutput, mFormat); 1720 lStatus = BAD_VALUE; 1721 goto Exit; 1722 } 1723 } 1724 } else { 1725 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1726 if (sampleRate > mSampleRate*2) { 1727 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1728 lStatus = BAD_VALUE; 1729 goto Exit; 1730 } 1731 } 1732 1733 lStatus = initCheck(); 1734 if (lStatus != NO_ERROR) { 1735 ALOGE("Audio driver not initialized."); 1736 goto Exit; 1737 } 1738 1739 { // scope for mLock 1740 Mutex::Autolock _l(mLock); 1741 1742 // all tracks in same audio session must share the same routing strategy otherwise 1743 // conflicts will happen when tracks are moved from one output to another by audio policy 1744 // manager 1745 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1746 for (size_t i = 0; i < mTracks.size(); ++i) { 1747 sp<Track> t = mTracks[i]; 1748 if (t != 0 && !t->isOutputTrack()) { 1749 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1750 if (sessionId == t->sessionId() && strategy != actual) { 1751 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1752 strategy, actual); 1753 lStatus = BAD_VALUE; 1754 goto Exit; 1755 } 1756 } 1757 } 1758 1759 if (!isTimed) { 1760 track = new Track(this, client, streamType, sampleRate, format, 1761 channelMask, frameCount, sharedBuffer, sessionId, flags); 1762 } else { 1763 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1764 channelMask, frameCount, sharedBuffer, sessionId); 1765 } 1766 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1767 lStatus = NO_MEMORY; 1768 goto Exit; 1769 } 1770 mTracks.add(track); 1771 1772 sp<EffectChain> chain = getEffectChain_l(sessionId); 1773 if (chain != 0) { 1774 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1775 track->setMainBuffer(chain->inBuffer()); 1776 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1777 chain->incTrackCnt(); 1778 } 1779 } 1780 1781#ifdef HAVE_REQUEST_PRIORITY 1782 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1783 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1784 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1785 // so ask activity manager to do this on our behalf 1786 int err = requestPriority(callingPid, tid, 1); 1787 if (err != 0) { 1788 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1789 1, callingPid, tid, err); 1790 } 1791 } 1792#endif 1793 1794 lStatus = NO_ERROR; 1795 1796Exit: 1797 if (status) { 1798 *status = lStatus; 1799 } 1800 return track; 1801} 1802 1803uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1804{ 1805 if (mFastMixer != NULL) { 1806 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1807 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1808 } 1809 return latency; 1810} 1811 1812uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1813{ 1814 return latency; 1815} 1816 1817uint32_t AudioFlinger::PlaybackThread::latency() const 1818{ 1819 Mutex::Autolock _l(mLock); 1820 if (initCheck() == NO_ERROR) { 1821 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1822 } else { 1823 return 0; 1824 } 1825} 1826 1827void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1828{ 1829 Mutex::Autolock _l(mLock); 1830 mMasterVolume = value; 1831} 1832 1833void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1834{ 1835 Mutex::Autolock _l(mLock); 1836 setMasterMute_l(muted); 1837} 1838 1839void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1840{ 1841 Mutex::Autolock _l(mLock); 1842 mStreamTypes[stream].volume = value; 1843} 1844 1845void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1846{ 1847 Mutex::Autolock _l(mLock); 1848 mStreamTypes[stream].mute = muted; 1849} 1850 1851float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1852{ 1853 Mutex::Autolock _l(mLock); 1854 return mStreamTypes[stream].volume; 1855} 1856 1857// addTrack_l() must be called with ThreadBase::mLock held 1858status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1859{ 1860 status_t status = ALREADY_EXISTS; 1861 1862 // set retry count for buffer fill 1863 track->mRetryCount = kMaxTrackStartupRetries; 1864 if (mActiveTracks.indexOf(track) < 0) { 1865 // the track is newly added, make sure it fills up all its 1866 // buffers before playing. This is to ensure the client will 1867 // effectively get the latency it requested. 1868 track->mFillingUpStatus = Track::FS_FILLING; 1869 track->mResetDone = false; 1870 track->mPresentationCompleteFrames = 0; 1871 mActiveTracks.add(track); 1872 if (track->mainBuffer() != mMixBuffer) { 1873 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1874 if (chain != 0) { 1875 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1876 chain->incActiveTrackCnt(); 1877 } 1878 } 1879 1880 status = NO_ERROR; 1881 } 1882 1883 ALOGV("mWaitWorkCV.broadcast"); 1884 mWaitWorkCV.broadcast(); 1885 1886 return status; 1887} 1888 1889// destroyTrack_l() must be called with ThreadBase::mLock held 1890void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1891{ 1892 track->mState = TrackBase::TERMINATED; 1893 // active tracks are removed by threadLoop() 1894 if (mActiveTracks.indexOf(track) < 0) { 1895 removeTrack_l(track); 1896 } 1897} 1898 1899void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1900{ 1901 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1902 mTracks.remove(track); 1903 deleteTrackName_l(track->name()); 1904 // redundant as track is about to be destroyed, for dumpsys only 1905 track->mName = -1; 1906 if (track->isFastTrack()) { 1907 int index = track->mFastIndex; 1908 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1909 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1910 mFastTrackAvailMask |= 1 << index; 1911 // redundant as track is about to be destroyed, for dumpsys only 1912 track->mFastIndex = -1; 1913 } 1914 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1915 if (chain != 0) { 1916 chain->decTrackCnt(); 1917 } 1918} 1919 1920String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1921{ 1922 String8 out_s8 = String8(""); 1923 char *s; 1924 1925 Mutex::Autolock _l(mLock); 1926 if (initCheck() != NO_ERROR) { 1927 return out_s8; 1928 } 1929 1930 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1931 out_s8 = String8(s); 1932 free(s); 1933 return out_s8; 1934} 1935 1936// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1937void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1938 AudioSystem::OutputDescriptor desc; 1939 void *param2 = NULL; 1940 1941 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1942 1943 switch (event) { 1944 case AudioSystem::OUTPUT_OPENED: 1945 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1946 desc.channels = mChannelMask; 1947 desc.samplingRate = mSampleRate; 1948 desc.format = mFormat; 1949 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1950 desc.latency = latency(); 1951 param2 = &desc; 1952 break; 1953 1954 case AudioSystem::STREAM_CONFIG_CHANGED: 1955 param2 = ¶m; 1956 case AudioSystem::OUTPUT_CLOSED: 1957 default: 1958 break; 1959 } 1960 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1961} 1962 1963void AudioFlinger::PlaybackThread::readOutputParameters() 1964{ 1965 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1966 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1967 mChannelCount = (uint16_t)popcount(mChannelMask); 1968 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1969 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1970 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1971 if (mFrameCount & 15) { 1972 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1973 mFrameCount); 1974 } 1975 1976 // Calculate size of normal mix buffer relative to the HAL output buffer size 1977 double multiplier = 1.0; 1978 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1979 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1980 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1981 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1982 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1983 maxNormalFrameCount = maxNormalFrameCount & ~15; 1984 if (maxNormalFrameCount < minNormalFrameCount) { 1985 maxNormalFrameCount = minNormalFrameCount; 1986 } 1987 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1988 if (multiplier <= 1.0) { 1989 multiplier = 1.0; 1990 } else if (multiplier <= 2.0) { 1991 if (2 * mFrameCount <= maxNormalFrameCount) { 1992 multiplier = 2.0; 1993 } else { 1994 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1995 } 1996 } else { 1997 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 1998 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 1999 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2000 // FIXME this rounding up should not be done if no HAL SRC 2001 uint32_t truncMult = (uint32_t) multiplier; 2002 if ((truncMult & 1)) { 2003 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2004 ++truncMult; 2005 } 2006 } 2007 multiplier = (double) truncMult; 2008 } 2009 } 2010 mNormalFrameCount = multiplier * mFrameCount; 2011 // round up to nearest 16 frames to satisfy AudioMixer 2012 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2013 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2014 2015 // FIXME - Current mixer implementation only supports stereo output: Always 2016 // Allocate a stereo buffer even if HW output is mono. 2017 delete[] mMixBuffer; 2018 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 2019 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 2020 2021 // force reconfiguration of effect chains and engines to take new buffer size and audio 2022 // parameters into account 2023 // Note that mLock is not held when readOutputParameters() is called from the constructor 2024 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2025 // matter. 2026 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2027 Vector< sp<EffectChain> > effectChains = mEffectChains; 2028 for (size_t i = 0; i < effectChains.size(); i ++) { 2029 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2030 } 2031} 2032 2033 2034status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2035{ 2036 if (halFrames == NULL || dspFrames == NULL) { 2037 return BAD_VALUE; 2038 } 2039 Mutex::Autolock _l(mLock); 2040 if (initCheck() != NO_ERROR) { 2041 return INVALID_OPERATION; 2042 } 2043 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2044 2045 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2046} 2047 2048uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2049{ 2050 Mutex::Autolock _l(mLock); 2051 uint32_t result = 0; 2052 if (getEffectChain_l(sessionId) != 0) { 2053 result = EFFECT_SESSION; 2054 } 2055 2056 for (size_t i = 0; i < mTracks.size(); ++i) { 2057 sp<Track> track = mTracks[i]; 2058 if (sessionId == track->sessionId() && 2059 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2060 result |= TRACK_SESSION; 2061 break; 2062 } 2063 } 2064 2065 return result; 2066} 2067 2068uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2069{ 2070 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2071 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2072 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2073 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2074 } 2075 for (size_t i = 0; i < mTracks.size(); i++) { 2076 sp<Track> track = mTracks[i]; 2077 if (sessionId == track->sessionId() && 2078 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2079 return AudioSystem::getStrategyForStream(track->streamType()); 2080 } 2081 } 2082 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2083} 2084 2085 2086AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2087{ 2088 Mutex::Autolock _l(mLock); 2089 return mOutput; 2090} 2091 2092AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2093{ 2094 Mutex::Autolock _l(mLock); 2095 AudioStreamOut *output = mOutput; 2096 mOutput = NULL; 2097 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2098 // must push a NULL and wait for ack 2099 mOutputSink.clear(); 2100 mPipeSink.clear(); 2101 mNormalSink.clear(); 2102 return output; 2103} 2104 2105// this method must always be called either with ThreadBase mLock held or inside the thread loop 2106audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2107{ 2108 if (mOutput == NULL) { 2109 return NULL; 2110 } 2111 return &mOutput->stream->common; 2112} 2113 2114uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2115{ 2116 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2117 // decoding and transfer time. So sleeping for half of the latency would likely cause 2118 // underruns 2119 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2120 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2121 } else { 2122 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2123 } 2124} 2125 2126status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2127{ 2128 if (!isValidSyncEvent(event)) { 2129 return BAD_VALUE; 2130 } 2131 2132 Mutex::Autolock _l(mLock); 2133 2134 for (size_t i = 0; i < mTracks.size(); ++i) { 2135 sp<Track> track = mTracks[i]; 2136 if (event->triggerSession() == track->sessionId()) { 2137 track->setSyncEvent(event); 2138 return NO_ERROR; 2139 } 2140 } 2141 2142 return NAME_NOT_FOUND; 2143} 2144 2145bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2146{ 2147 switch (event->type()) { 2148 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2149 return true; 2150 default: 2151 break; 2152 } 2153 return false; 2154} 2155 2156void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2157{ 2158 size_t count = tracksToRemove.size(); 2159 if (CC_UNLIKELY(count)) { 2160 for (size_t i = 0 ; i < count ; i++) { 2161 const sp<Track>& track = tracksToRemove.itemAt(i); 2162 if ((track->sharedBuffer() != 0) && 2163 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2164 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2165 } 2166 } 2167 } 2168 2169} 2170 2171// ---------------------------------------------------------------------------- 2172 2173AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2174 audio_io_handle_t id, uint32_t device, type_t type) 2175 : PlaybackThread(audioFlinger, output, id, device, type), 2176 // mAudioMixer below 2177#ifdef SOAKER 2178 mSoaker(NULL), 2179#endif 2180 // mFastMixer below 2181 mFastMixerFutex(0) 2182 // mOutputSink below 2183 // mPipeSink below 2184 // mNormalSink below 2185{ 2186 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2187 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2188 "mFrameCount=%d, mNormalFrameCount=%d", 2189 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2190 mNormalFrameCount); 2191 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2192 2193 // FIXME - Current mixer implementation only supports stereo output 2194 if (mChannelCount == 1) { 2195 ALOGE("Invalid audio hardware channel count"); 2196 } 2197 2198 // create an NBAIO sink for the HAL output stream, and negotiate 2199 mOutputSink = new AudioStreamOutSink(output->stream); 2200 size_t numCounterOffers = 0; 2201 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2202 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2203 ALOG_ASSERT(index == 0); 2204 2205 // initialize fast mixer depending on configuration 2206 bool initFastMixer; 2207 switch (kUseFastMixer) { 2208 case FastMixer_Never: 2209 initFastMixer = false; 2210 break; 2211 case FastMixer_Always: 2212 initFastMixer = true; 2213 break; 2214 case FastMixer_Static: 2215 case FastMixer_Dynamic: 2216 initFastMixer = mFrameCount < mNormalFrameCount; 2217 break; 2218 } 2219 if (initFastMixer) { 2220 2221 // create a MonoPipe to connect our submix to FastMixer 2222 NBAIO_Format format = mOutputSink->format(); 2223 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2224 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2225 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2226 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2227 const NBAIO_Format offers[1] = {format}; 2228 size_t numCounterOffers = 0; 2229 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2230 ALOG_ASSERT(index == 0); 2231 mPipeSink = monoPipe; 2232 2233#ifdef TEE_SINK_FRAMES 2234 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2235 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2236 numCounterOffers = 0; 2237 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2238 ALOG_ASSERT(index == 0); 2239 mTeeSink = teeSink; 2240 PipeReader *teeSource = new PipeReader(*teeSink); 2241 numCounterOffers = 0; 2242 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2243 ALOG_ASSERT(index == 0); 2244 mTeeSource = teeSource; 2245#endif 2246 2247#ifdef SOAKER 2248 // create a soaker as workaround for governor issues 2249 mSoaker = new Soaker(); 2250 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2251 mSoaker->run("Soaker", PRIORITY_LOWEST); 2252#endif 2253 2254 // create fast mixer and configure it initially with just one fast track for our submix 2255 mFastMixer = new FastMixer(); 2256 FastMixerStateQueue *sq = mFastMixer->sq(); 2257 FastMixerState *state = sq->begin(); 2258 FastTrack *fastTrack = &state->mFastTracks[0]; 2259 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2260 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2261 fastTrack->mVolumeProvider = NULL; 2262 fastTrack->mGeneration++; 2263 state->mFastTracksGen++; 2264 state->mTrackMask = 1; 2265 // fast mixer will use the HAL output sink 2266 state->mOutputSink = mOutputSink.get(); 2267 state->mOutputSinkGen++; 2268 state->mFrameCount = mFrameCount; 2269 state->mCommand = FastMixerState::COLD_IDLE; 2270 // already done in constructor initialization list 2271 //mFastMixerFutex = 0; 2272 state->mColdFutexAddr = &mFastMixerFutex; 2273 state->mColdGen++; 2274 state->mDumpState = &mFastMixerDumpState; 2275 state->mTeeSink = mTeeSink.get(); 2276 sq->end(); 2277 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2278 2279 // start the fast mixer 2280 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2281#ifdef HAVE_REQUEST_PRIORITY 2282 pid_t tid = mFastMixer->getTid(); 2283 int err = requestPriority(getpid_cached, tid, 2); 2284 if (err != 0) { 2285 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2286 2, getpid_cached, tid, err); 2287 } 2288#endif 2289 2290 } else { 2291 mFastMixer = NULL; 2292 } 2293 2294 switch (kUseFastMixer) { 2295 case FastMixer_Never: 2296 case FastMixer_Dynamic: 2297 mNormalSink = mOutputSink; 2298 break; 2299 case FastMixer_Always: 2300 mNormalSink = mPipeSink; 2301 break; 2302 case FastMixer_Static: 2303 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2304 break; 2305 } 2306} 2307 2308AudioFlinger::MixerThread::~MixerThread() 2309{ 2310 if (mFastMixer != NULL) { 2311 FastMixerStateQueue *sq = mFastMixer->sq(); 2312 FastMixerState *state = sq->begin(); 2313 if (state->mCommand == FastMixerState::COLD_IDLE) { 2314 int32_t old = android_atomic_inc(&mFastMixerFutex); 2315 if (old == -1) { 2316 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2317 } 2318 } 2319 state->mCommand = FastMixerState::EXIT; 2320 sq->end(); 2321 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2322 mFastMixer->join(); 2323 // Though the fast mixer thread has exited, it's state queue is still valid. 2324 // We'll use that extract the final state which contains one remaining fast track 2325 // corresponding to our sub-mix. 2326 state = sq->begin(); 2327 ALOG_ASSERT(state->mTrackMask == 1); 2328 FastTrack *fastTrack = &state->mFastTracks[0]; 2329 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2330 delete fastTrack->mBufferProvider; 2331 sq->end(false /*didModify*/); 2332 delete mFastMixer; 2333#ifdef SOAKER 2334 if (mSoaker != NULL) { 2335 mSoaker->requestExitAndWait(); 2336 } 2337 delete mSoaker; 2338#endif 2339 } 2340 delete mAudioMixer; 2341} 2342 2343class CpuStats { 2344public: 2345 CpuStats(); 2346 void sample(const String8 &title); 2347#ifdef DEBUG_CPU_USAGE 2348private: 2349 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2350 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2351 2352 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2353 2354 int mCpuNum; // thread's current CPU number 2355 int mCpukHz; // frequency of thread's current CPU in kHz 2356#endif 2357}; 2358 2359CpuStats::CpuStats() 2360#ifdef DEBUG_CPU_USAGE 2361 : mCpuNum(-1), mCpukHz(-1) 2362#endif 2363{ 2364} 2365 2366void CpuStats::sample(const String8 &title) { 2367#ifdef DEBUG_CPU_USAGE 2368 // get current thread's delta CPU time in wall clock ns 2369 double wcNs; 2370 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2371 2372 // record sample for wall clock statistics 2373 if (valid) { 2374 mWcStats.sample(wcNs); 2375 } 2376 2377 // get the current CPU number 2378 int cpuNum = sched_getcpu(); 2379 2380 // get the current CPU frequency in kHz 2381 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2382 2383 // check if either CPU number or frequency changed 2384 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2385 mCpuNum = cpuNum; 2386 mCpukHz = cpukHz; 2387 // ignore sample for purposes of cycles 2388 valid = false; 2389 } 2390 2391 // if no change in CPU number or frequency, then record sample for cycle statistics 2392 if (valid && mCpukHz > 0) { 2393 double cycles = wcNs * cpukHz * 0.000001; 2394 mHzStats.sample(cycles); 2395 } 2396 2397 unsigned n = mWcStats.n(); 2398 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2399 if ((n & 127) == 1) { 2400 long long elapsed = mCpuUsage.elapsed(); 2401 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2402 double perLoop = elapsed / (double) n; 2403 double perLoop100 = perLoop * 0.01; 2404 double perLoop1k = perLoop * 0.001; 2405 double mean = mWcStats.mean(); 2406 double stddev = mWcStats.stddev(); 2407 double minimum = mWcStats.minimum(); 2408 double maximum = mWcStats.maximum(); 2409 double meanCycles = mHzStats.mean(); 2410 double stddevCycles = mHzStats.stddev(); 2411 double minCycles = mHzStats.minimum(); 2412 double maxCycles = mHzStats.maximum(); 2413 mCpuUsage.resetElapsed(); 2414 mWcStats.reset(); 2415 mHzStats.reset(); 2416 ALOGD("CPU usage for %s over past %.1f secs\n" 2417 " (%u mixer loops at %.1f mean ms per loop):\n" 2418 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2419 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2420 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2421 title.string(), 2422 elapsed * .000000001, n, perLoop * .000001, 2423 mean * .001, 2424 stddev * .001, 2425 minimum * .001, 2426 maximum * .001, 2427 mean / perLoop100, 2428 stddev / perLoop100, 2429 minimum / perLoop100, 2430 maximum / perLoop100, 2431 meanCycles / perLoop1k, 2432 stddevCycles / perLoop1k, 2433 minCycles / perLoop1k, 2434 maxCycles / perLoop1k); 2435 2436 } 2437 } 2438#endif 2439}; 2440 2441void AudioFlinger::PlaybackThread::checkSilentMode_l() 2442{ 2443 if (!mMasterMute) { 2444 char value[PROPERTY_VALUE_MAX]; 2445 if (property_get("ro.audio.silent", value, "0") > 0) { 2446 char *endptr; 2447 unsigned long ul = strtoul(value, &endptr, 0); 2448 if (*endptr == '\0' && ul != 0) { 2449 ALOGD("Silence is golden"); 2450 // The setprop command will not allow a property to be changed after 2451 // the first time it is set, so we don't have to worry about un-muting. 2452 setMasterMute_l(true); 2453 } 2454 } 2455 } 2456} 2457 2458bool AudioFlinger::PlaybackThread::threadLoop() 2459{ 2460 Vector< sp<Track> > tracksToRemove; 2461 2462 standbyTime = systemTime(); 2463 2464 // MIXER 2465 nsecs_t lastWarning = 0; 2466if (mType == MIXER) { 2467 longStandbyExit = false; 2468} 2469 2470 // DUPLICATING 2471 // FIXME could this be made local to while loop? 2472 writeFrames = 0; 2473 2474 cacheParameters_l(); 2475 sleepTime = idleSleepTime; 2476 2477if (mType == MIXER) { 2478 sleepTimeShift = 0; 2479} 2480 2481 CpuStats cpuStats; 2482 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2483 2484 acquireWakeLock(); 2485 2486 while (!exitPending()) 2487 { 2488 cpuStats.sample(myName); 2489 2490 Vector< sp<EffectChain> > effectChains; 2491 2492 processConfigEvents(); 2493 2494 { // scope for mLock 2495 2496 Mutex::Autolock _l(mLock); 2497 2498 if (checkForNewParameters_l()) { 2499 cacheParameters_l(); 2500 } 2501 2502 saveOutputTracks(); 2503 2504 // put audio hardware into standby after short delay 2505 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2506 mSuspended > 0)) { 2507 if (!mStandby) { 2508 2509 threadLoop_standby(); 2510 2511 mStandby = true; 2512 mBytesWritten = 0; 2513 } 2514 2515 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2516 // we're about to wait, flush the binder command buffer 2517 IPCThreadState::self()->flushCommands(); 2518 2519 clearOutputTracks(); 2520 2521 if (exitPending()) break; 2522 2523 releaseWakeLock_l(); 2524 // wait until we have something to do... 2525 ALOGV("%s going to sleep", myName.string()); 2526 mWaitWorkCV.wait(mLock); 2527 ALOGV("%s waking up", myName.string()); 2528 acquireWakeLock_l(); 2529 2530 mMixerStatus = MIXER_IDLE; 2531 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2532 2533 checkSilentMode_l(); 2534 2535 standbyTime = systemTime() + standbyDelay; 2536 sleepTime = idleSleepTime; 2537 if (mType == MIXER) { 2538 sleepTimeShift = 0; 2539 } 2540 2541 continue; 2542 } 2543 } 2544 2545 // mMixerStatusIgnoringFastTracks is also updated internally 2546 mMixerStatus = prepareTracks_l(&tracksToRemove); 2547 2548 // prevent any changes in effect chain list and in each effect chain 2549 // during mixing and effect process as the audio buffers could be deleted 2550 // or modified if an effect is created or deleted 2551 lockEffectChains_l(effectChains); 2552 } 2553 2554 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2555 threadLoop_mix(); 2556 } else { 2557 threadLoop_sleepTime(); 2558 } 2559 2560 if (mSuspended > 0) { 2561 sleepTime = suspendSleepTimeUs(); 2562 } 2563 2564 // only process effects if we're going to write 2565 if (sleepTime == 0) { 2566 for (size_t i = 0; i < effectChains.size(); i ++) { 2567 effectChains[i]->process_l(); 2568 } 2569 } 2570 2571 // enable changes in effect chain 2572 unlockEffectChains(effectChains); 2573 2574 // sleepTime == 0 means we must write to audio hardware 2575 if (sleepTime == 0) { 2576 2577 threadLoop_write(); 2578 2579if (mType == MIXER) { 2580 // write blocked detection 2581 nsecs_t now = systemTime(); 2582 nsecs_t delta = now - mLastWriteTime; 2583 if (!mStandby && delta > maxPeriod) { 2584 mNumDelayedWrites++; 2585 if ((now - lastWarning) > kWarningThrottleNs) { 2586#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2587 ScopedTrace st(ATRACE_TAG, "underrun"); 2588#endif 2589 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2590 ns2ms(delta), mNumDelayedWrites, this); 2591 lastWarning = now; 2592 } 2593 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2594 // a different threshold. Or completely removed for what it is worth anyway... 2595 if (mStandby) { 2596 longStandbyExit = true; 2597 } 2598 } 2599} 2600 2601 mStandby = false; 2602 } else { 2603 usleep(sleepTime); 2604 } 2605 2606 // Finally let go of removed track(s), without the lock held 2607 // since we can't guarantee the destructors won't acquire that 2608 // same lock. This will also mutate and push a new fast mixer state. 2609 threadLoop_removeTracks(tracksToRemove); 2610 tracksToRemove.clear(); 2611 2612 // FIXME I don't understand the need for this here; 2613 // it was in the original code but maybe the 2614 // assignment in saveOutputTracks() makes this unnecessary? 2615 clearOutputTracks(); 2616 2617 // Effect chains will be actually deleted here if they were removed from 2618 // mEffectChains list during mixing or effects processing 2619 effectChains.clear(); 2620 2621 // FIXME Note that the above .clear() is no longer necessary since effectChains 2622 // is now local to this block, but will keep it for now (at least until merge done). 2623 } 2624 2625if (mType == MIXER || mType == DIRECT) { 2626 // put output stream into standby mode 2627 if (!mStandby) { 2628 mOutput->stream->common.standby(&mOutput->stream->common); 2629 } 2630} 2631if (mType == DUPLICATING) { 2632 // for DuplicatingThread, standby mode is handled by the outputTracks 2633} 2634 2635 releaseWakeLock(); 2636 2637 ALOGV("Thread %p type %d exiting", this, mType); 2638 return false; 2639} 2640 2641void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2642{ 2643 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2644} 2645 2646void AudioFlinger::MixerThread::threadLoop_write() 2647{ 2648 // FIXME we should only do one push per cycle; confirm this is true 2649 // Start the fast mixer if it's not already running 2650 if (mFastMixer != NULL) { 2651 FastMixerStateQueue *sq = mFastMixer->sq(); 2652 FastMixerState *state = sq->begin(); 2653 if (state->mCommand != FastMixerState::MIX_WRITE && 2654 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2655 if (state->mCommand == FastMixerState::COLD_IDLE) { 2656 int32_t old = android_atomic_inc(&mFastMixerFutex); 2657 if (old == -1) { 2658 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2659 } 2660 } 2661 state->mCommand = FastMixerState::MIX_WRITE; 2662 sq->end(); 2663 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2664 if (kUseFastMixer == FastMixer_Dynamic) { 2665 mNormalSink = mPipeSink; 2666 } 2667 } else { 2668 sq->end(false /*didModify*/); 2669 } 2670 } 2671 PlaybackThread::threadLoop_write(); 2672} 2673 2674// shared by MIXER and DIRECT, overridden by DUPLICATING 2675void AudioFlinger::PlaybackThread::threadLoop_write() 2676{ 2677 // FIXME rewrite to reduce number of system calls 2678 mLastWriteTime = systemTime(); 2679 mInWrite = true; 2680 2681#define mBitShift 2 // FIXME 2682 size_t count = mixBufferSize >> mBitShift; 2683#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2684 Tracer::traceBegin(ATRACE_TAG, "write"); 2685#endif 2686 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2687#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2688 Tracer::traceEnd(ATRACE_TAG); 2689#endif 2690 if (framesWritten > 0) { 2691 size_t bytesWritten = framesWritten << mBitShift; 2692 mBytesWritten += bytesWritten; 2693 } 2694 2695 mNumWrites++; 2696 mInWrite = false; 2697} 2698 2699void AudioFlinger::MixerThread::threadLoop_standby() 2700{ 2701 // Idle the fast mixer if it's currently running 2702 if (mFastMixer != NULL) { 2703 FastMixerStateQueue *sq = mFastMixer->sq(); 2704 FastMixerState *state = sq->begin(); 2705 if (!(state->mCommand & FastMixerState::IDLE)) { 2706 state->mCommand = FastMixerState::COLD_IDLE; 2707 state->mColdFutexAddr = &mFastMixerFutex; 2708 state->mColdGen++; 2709 mFastMixerFutex = 0; 2710 sq->end(); 2711 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2712 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2713 if (kUseFastMixer == FastMixer_Dynamic) { 2714 mNormalSink = mOutputSink; 2715 } 2716 } else { 2717 sq->end(false /*didModify*/); 2718 } 2719 } 2720 PlaybackThread::threadLoop_standby(); 2721} 2722 2723// shared by MIXER and DIRECT, overridden by DUPLICATING 2724void AudioFlinger::PlaybackThread::threadLoop_standby() 2725{ 2726 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2727 mOutput->stream->common.standby(&mOutput->stream->common); 2728} 2729 2730void AudioFlinger::MixerThread::threadLoop_mix() 2731{ 2732 // obtain the presentation timestamp of the next output buffer 2733 int64_t pts; 2734 status_t status = INVALID_OPERATION; 2735 2736 if (NULL != mOutput->stream->get_next_write_timestamp) { 2737 status = mOutput->stream->get_next_write_timestamp( 2738 mOutput->stream, &pts); 2739 } 2740 2741 if (status != NO_ERROR) { 2742 pts = AudioBufferProvider::kInvalidPTS; 2743 } 2744 2745 // mix buffers... 2746 mAudioMixer->process(pts); 2747 // increase sleep time progressively when application underrun condition clears. 2748 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2749 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2750 // such that we would underrun the audio HAL. 2751 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2752 sleepTimeShift--; 2753 } 2754 sleepTime = 0; 2755 standbyTime = systemTime() + standbyDelay; 2756 //TODO: delay standby when effects have a tail 2757} 2758 2759void AudioFlinger::MixerThread::threadLoop_sleepTime() 2760{ 2761 // If no tracks are ready, sleep once for the duration of an output 2762 // buffer size, then write 0s to the output 2763 if (sleepTime == 0) { 2764 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2765 sleepTime = activeSleepTime >> sleepTimeShift; 2766 if (sleepTime < kMinThreadSleepTimeUs) { 2767 sleepTime = kMinThreadSleepTimeUs; 2768 } 2769 // reduce sleep time in case of consecutive application underruns to avoid 2770 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2771 // duration we would end up writing less data than needed by the audio HAL if 2772 // the condition persists. 2773 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2774 sleepTimeShift++; 2775 } 2776 } else { 2777 sleepTime = idleSleepTime; 2778 } 2779 } else if (mBytesWritten != 0 || 2780 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2781 memset (mMixBuffer, 0, mixBufferSize); 2782 sleepTime = 0; 2783 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2784 } 2785 // TODO add standby time extension fct of effect tail 2786} 2787 2788// prepareTracks_l() must be called with ThreadBase::mLock held 2789AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2790 Vector< sp<Track> > *tracksToRemove) 2791{ 2792 2793 mixer_state mixerStatus = MIXER_IDLE; 2794 // find out which tracks need to be processed 2795 size_t count = mActiveTracks.size(); 2796 size_t mixedTracks = 0; 2797 size_t tracksWithEffect = 0; 2798 // counts only _active_ fast tracks 2799 size_t fastTracks = 0; 2800 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2801 2802 float masterVolume = mMasterVolume; 2803 bool masterMute = mMasterMute; 2804 2805 if (masterMute) { 2806 masterVolume = 0; 2807 } 2808 // Delegate master volume control to effect in output mix effect chain if needed 2809 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2810 if (chain != 0) { 2811 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2812 chain->setVolume_l(&v, &v); 2813 masterVolume = (float)((v + (1 << 23)) >> 24); 2814 chain.clear(); 2815 } 2816 2817 // prepare a new state to push 2818 FastMixerStateQueue *sq = NULL; 2819 FastMixerState *state = NULL; 2820 bool didModify = false; 2821 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2822 if (mFastMixer != NULL) { 2823 sq = mFastMixer->sq(); 2824 state = sq->begin(); 2825 } 2826 2827 for (size_t i=0 ; i<count ; i++) { 2828 sp<Track> t = mActiveTracks[i].promote(); 2829 if (t == 0) continue; 2830 2831 // this const just means the local variable doesn't change 2832 Track* const track = t.get(); 2833 2834 // process fast tracks 2835 if (track->isFastTrack()) { 2836 2837 // It's theoretically possible (though unlikely) for a fast track to be created 2838 // and then removed within the same normal mix cycle. This is not a problem, as 2839 // the track never becomes active so it's fast mixer slot is never touched. 2840 // The converse, of removing an (active) track and then creating a new track 2841 // at the identical fast mixer slot within the same normal mix cycle, 2842 // is impossible because the slot isn't marked available until the end of each cycle. 2843 int j = track->mFastIndex; 2844 FastTrack *fastTrack = &state->mFastTracks[j]; 2845 2846 // Determine whether the track is currently in underrun condition, 2847 // and whether it had a recent underrun. 2848 FastTrackUnderruns underruns = mFastMixerDumpState.mTracks[j].mUnderruns; 2849 uint32_t recentFull = (underruns.mBitFields.mFull - 2850 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2851 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2852 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2853 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2854 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2855 uint32_t recentUnderruns = recentPartial + recentEmpty; 2856 track->mObservedUnderruns = underruns; 2857 // don't count underruns that occur while stopping or pausing 2858 // or stopped which can occur when flush() is called while active 2859 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2860 track->mUnderrunCount += recentUnderruns; 2861 } 2862 2863 // This is similar to the state machine for normal tracks, 2864 // with a few modifications for fast tracks. 2865 bool isActive = true; 2866 switch (track->mState) { 2867 case TrackBase::STOPPING_1: 2868 // track stays active in STOPPING_1 state until first underrun 2869 if (recentUnderruns > 0) { 2870 track->mState = TrackBase::STOPPING_2; 2871 } 2872 break; 2873 case TrackBase::PAUSING: 2874 // ramp down is not yet implemented 2875 track->setPaused(); 2876 break; 2877 case TrackBase::RESUMING: 2878 // ramp up is not yet implemented 2879 track->mState = TrackBase::ACTIVE; 2880 break; 2881 case TrackBase::ACTIVE: 2882 if (recentFull > 0 || recentPartial > 0) { 2883 // track has provided at least some frames recently: reset retry count 2884 track->mRetryCount = kMaxTrackRetries; 2885 } 2886 if (recentUnderruns == 0) { 2887 // no recent underruns: stay active 2888 break; 2889 } 2890 // there has recently been an underrun of some kind 2891 if (track->sharedBuffer() == 0) { 2892 // were any of the recent underruns "empty" (no frames available)? 2893 if (recentEmpty == 0) { 2894 // no, then ignore the partial underruns as they are allowed indefinitely 2895 break; 2896 } 2897 // there has recently been an "empty" underrun: decrement the retry counter 2898 if (--(track->mRetryCount) > 0) { 2899 break; 2900 } 2901 // indicate to client process that the track was disabled because of underrun; 2902 // it will then automatically call start() when data is available 2903 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2904 // remove from active list, but state remains ACTIVE [confusing but true] 2905 isActive = false; 2906 break; 2907 } 2908 // fall through 2909 case TrackBase::STOPPING_2: 2910 case TrackBase::PAUSED: 2911 case TrackBase::TERMINATED: 2912 case TrackBase::STOPPED: 2913 case TrackBase::FLUSHED: // flush() while active 2914 // Check for presentation complete if track is inactive 2915 // We have consumed all the buffers of this track. 2916 // This would be incomplete if we auto-paused on underrun 2917 { 2918 size_t audioHALFrames = 2919 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2920 size_t framesWritten = 2921 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2922 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2923 // track stays in active list until presentation is complete 2924 break; 2925 } 2926 } 2927 if (track->isStopping_2()) { 2928 track->mState = TrackBase::STOPPED; 2929 } 2930 if (track->isStopped()) { 2931 // Can't reset directly, as fast mixer is still polling this track 2932 // track->reset(); 2933 // So instead mark this track as needing to be reset after push with ack 2934 resetMask |= 1 << i; 2935 } 2936 isActive = false; 2937 break; 2938 case TrackBase::IDLE: 2939 default: 2940 LOG_FATAL("unexpected track state %d", track->mState); 2941 } 2942 2943 if (isActive) { 2944 // was it previously inactive? 2945 if (!(state->mTrackMask & (1 << j))) { 2946 ExtendedAudioBufferProvider *eabp = track; 2947 VolumeProvider *vp = track; 2948 fastTrack->mBufferProvider = eabp; 2949 fastTrack->mVolumeProvider = vp; 2950 fastTrack->mSampleRate = track->mSampleRate; 2951 fastTrack->mChannelMask = track->mChannelMask; 2952 fastTrack->mGeneration++; 2953 state->mTrackMask |= 1 << j; 2954 didModify = true; 2955 // no acknowledgement required for newly active tracks 2956 } 2957 // cache the combined master volume and stream type volume for fast mixer; this 2958 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2959 track->mCachedVolume = track->isMuted() ? 2960 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2961 ++fastTracks; 2962 } else { 2963 // was it previously active? 2964 if (state->mTrackMask & (1 << j)) { 2965 fastTrack->mBufferProvider = NULL; 2966 fastTrack->mGeneration++; 2967 state->mTrackMask &= ~(1 << j); 2968 didModify = true; 2969 // If any fast tracks were removed, we must wait for acknowledgement 2970 // because we're about to decrement the last sp<> on those tracks. 2971 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2972 } else { 2973 LOG_FATAL("fast track %d should have been active", j); 2974 } 2975 tracksToRemove->add(track); 2976 // Avoids a misleading display in dumpsys 2977 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2978 } 2979 continue; 2980 } 2981 2982 { // local variable scope to avoid goto warning 2983 2984 audio_track_cblk_t* cblk = track->cblk(); 2985 2986 // The first time a track is added we wait 2987 // for all its buffers to be filled before processing it 2988 int name = track->name(); 2989 // make sure that we have enough frames to mix one full buffer. 2990 // enforce this condition only once to enable draining the buffer in case the client 2991 // app does not call stop() and relies on underrun to stop: 2992 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2993 // during last round 2994 uint32_t minFrames = 1; 2995 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2996 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2997 if (t->sampleRate() == (int)mSampleRate) { 2998 minFrames = mNormalFrameCount; 2999 } else { 3000 // +1 for rounding and +1 for additional sample needed for interpolation 3001 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3002 // add frames already consumed but not yet released by the resampler 3003 // because cblk->framesReady() will include these frames 3004 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3005 // the minimum track buffer size is normally twice the number of frames necessary 3006 // to fill one buffer and the resampler should not leave more than one buffer worth 3007 // of unreleased frames after each pass, but just in case... 3008 ALOG_ASSERT(minFrames <= cblk->frameCount); 3009 } 3010 } 3011 if ((track->framesReady() >= minFrames) && track->isReady() && 3012 !track->isPaused() && !track->isTerminated()) 3013 { 3014 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3015 3016 mixedTracks++; 3017 3018 // track->mainBuffer() != mMixBuffer means there is an effect chain 3019 // connected to the track 3020 chain.clear(); 3021 if (track->mainBuffer() != mMixBuffer) { 3022 chain = getEffectChain_l(track->sessionId()); 3023 // Delegate volume control to effect in track effect chain if needed 3024 if (chain != 0) { 3025 tracksWithEffect++; 3026 } else { 3027 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3028 name, track->sessionId()); 3029 } 3030 } 3031 3032 3033 int param = AudioMixer::VOLUME; 3034 if (track->mFillingUpStatus == Track::FS_FILLED) { 3035 // no ramp for the first volume setting 3036 track->mFillingUpStatus = Track::FS_ACTIVE; 3037 if (track->mState == TrackBase::RESUMING) { 3038 track->mState = TrackBase::ACTIVE; 3039 param = AudioMixer::RAMP_VOLUME; 3040 } 3041 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3042 } else if (cblk->server != 0) { 3043 // If the track is stopped before the first frame was mixed, 3044 // do not apply ramp 3045 param = AudioMixer::RAMP_VOLUME; 3046 } 3047 3048 // compute volume for this track 3049 uint32_t vl, vr, va; 3050 if (track->isMuted() || track->isPausing() || 3051 mStreamTypes[track->streamType()].mute) { 3052 vl = vr = va = 0; 3053 if (track->isPausing()) { 3054 track->setPaused(); 3055 } 3056 } else { 3057 3058 // read original volumes with volume control 3059 float typeVolume = mStreamTypes[track->streamType()].volume; 3060 float v = masterVolume * typeVolume; 3061 uint32_t vlr = cblk->getVolumeLR(); 3062 vl = vlr & 0xFFFF; 3063 vr = vlr >> 16; 3064 // track volumes come from shared memory, so can't be trusted and must be clamped 3065 if (vl > MAX_GAIN_INT) { 3066 ALOGV("Track left volume out of range: %04X", vl); 3067 vl = MAX_GAIN_INT; 3068 } 3069 if (vr > MAX_GAIN_INT) { 3070 ALOGV("Track right volume out of range: %04X", vr); 3071 vr = MAX_GAIN_INT; 3072 } 3073 // now apply the master volume and stream type volume 3074 vl = (uint32_t)(v * vl) << 12; 3075 vr = (uint32_t)(v * vr) << 12; 3076 // assuming master volume and stream type volume each go up to 1.0, 3077 // vl and vr are now in 8.24 format 3078 3079 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3080 // send level comes from shared memory and so may be corrupt 3081 if (sendLevel > MAX_GAIN_INT) { 3082 ALOGV("Track send level out of range: %04X", sendLevel); 3083 sendLevel = MAX_GAIN_INT; 3084 } 3085 va = (uint32_t)(v * sendLevel); 3086 } 3087 // Delegate volume control to effect in track effect chain if needed 3088 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3089 // Do not ramp volume if volume is controlled by effect 3090 param = AudioMixer::VOLUME; 3091 track->mHasVolumeController = true; 3092 } else { 3093 // force no volume ramp when volume controller was just disabled or removed 3094 // from effect chain to avoid volume spike 3095 if (track->mHasVolumeController) { 3096 param = AudioMixer::VOLUME; 3097 } 3098 track->mHasVolumeController = false; 3099 } 3100 3101 // Convert volumes from 8.24 to 4.12 format 3102 // This additional clamping is needed in case chain->setVolume_l() overshot 3103 vl = (vl + (1 << 11)) >> 12; 3104 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3105 vr = (vr + (1 << 11)) >> 12; 3106 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3107 3108 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3109 3110 // XXX: these things DON'T need to be done each time 3111 mAudioMixer->setBufferProvider(name, track); 3112 mAudioMixer->enable(name); 3113 3114 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3115 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3116 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3117 mAudioMixer->setParameter( 3118 name, 3119 AudioMixer::TRACK, 3120 AudioMixer::FORMAT, (void *)track->format()); 3121 mAudioMixer->setParameter( 3122 name, 3123 AudioMixer::TRACK, 3124 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3125 mAudioMixer->setParameter( 3126 name, 3127 AudioMixer::RESAMPLE, 3128 AudioMixer::SAMPLE_RATE, 3129 (void *)(cblk->sampleRate)); 3130 mAudioMixer->setParameter( 3131 name, 3132 AudioMixer::TRACK, 3133 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3134 mAudioMixer->setParameter( 3135 name, 3136 AudioMixer::TRACK, 3137 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3138 3139 // reset retry count 3140 track->mRetryCount = kMaxTrackRetries; 3141 3142 // If one track is ready, set the mixer ready if: 3143 // - the mixer was not ready during previous round OR 3144 // - no other track is not ready 3145 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3146 mixerStatus != MIXER_TRACKS_ENABLED) { 3147 mixerStatus = MIXER_TRACKS_READY; 3148 } 3149 } else { 3150 // clear effect chain input buffer if an active track underruns to avoid sending 3151 // previous audio buffer again to effects 3152 chain = getEffectChain_l(track->sessionId()); 3153 if (chain != 0) { 3154 chain->clearInputBuffer(); 3155 } 3156 3157 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3158 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3159 track->isStopped() || track->isPaused()) { 3160 // We have consumed all the buffers of this track. 3161 // Remove it from the list of active tracks. 3162 // TODO: use actual buffer filling status instead of latency when available from 3163 // audio HAL 3164 size_t audioHALFrames = 3165 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3166 size_t framesWritten = 3167 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3168 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3169 if (track->isStopped()) { 3170 track->reset(); 3171 } 3172 tracksToRemove->add(track); 3173 } 3174 } else { 3175 // No buffers for this track. Give it a few chances to 3176 // fill a buffer, then remove it from active list. 3177 if (--(track->mRetryCount) <= 0) { 3178 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3179 tracksToRemove->add(track); 3180 // indicate to client process that the track was disabled because of underrun; 3181 // it will then automatically call start() when data is available 3182 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3183 // If one track is not ready, mark the mixer also not ready if: 3184 // - the mixer was ready during previous round OR 3185 // - no other track is ready 3186 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3187 mixerStatus != MIXER_TRACKS_READY) { 3188 mixerStatus = MIXER_TRACKS_ENABLED; 3189 } 3190 } 3191 mAudioMixer->disable(name); 3192 } 3193 3194 } // local variable scope to avoid goto warning 3195track_is_ready: ; 3196 3197 } 3198 3199 // Push the new FastMixer state if necessary 3200 if (didModify) { 3201 state->mFastTracksGen++; 3202 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3203 if (kUseFastMixer == FastMixer_Dynamic && 3204 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3205 state->mCommand = FastMixerState::COLD_IDLE; 3206 state->mColdFutexAddr = &mFastMixerFutex; 3207 state->mColdGen++; 3208 mFastMixerFutex = 0; 3209 if (kUseFastMixer == FastMixer_Dynamic) { 3210 mNormalSink = mOutputSink; 3211 } 3212 // If we go into cold idle, need to wait for acknowledgement 3213 // so that fast mixer stops doing I/O. 3214 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3215 } 3216 sq->end(); 3217 } 3218 if (sq != NULL) { 3219 sq->end(didModify); 3220 sq->push(block); 3221 } 3222 3223 // Now perform the deferred reset on fast tracks that have stopped 3224 while (resetMask != 0) { 3225 size_t i = __builtin_ctz(resetMask); 3226 ALOG_ASSERT(i < count); 3227 resetMask &= ~(1 << i); 3228 sp<Track> t = mActiveTracks[i].promote(); 3229 if (t == 0) continue; 3230 Track* track = t.get(); 3231 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3232 track->reset(); 3233 } 3234 3235 // remove all the tracks that need to be... 3236 count = tracksToRemove->size(); 3237 if (CC_UNLIKELY(count)) { 3238 for (size_t i=0 ; i<count ; i++) { 3239 const sp<Track>& track = tracksToRemove->itemAt(i); 3240 mActiveTracks.remove(track); 3241 if (track->mainBuffer() != mMixBuffer) { 3242 chain = getEffectChain_l(track->sessionId()); 3243 if (chain != 0) { 3244 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3245 chain->decActiveTrackCnt(); 3246 } 3247 } 3248 if (track->isTerminated()) { 3249 removeTrack_l(track); 3250 } 3251 } 3252 } 3253 3254 // mix buffer must be cleared if all tracks are connected to an 3255 // effect chain as in this case the mixer will not write to 3256 // mix buffer and track effects will accumulate into it 3257 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3258 // FIXME as a performance optimization, should remember previous zero status 3259 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3260 } 3261 3262 // if any fast tracks, then status is ready 3263 mMixerStatusIgnoringFastTracks = mixerStatus; 3264 if (fastTracks > 0) { 3265 mixerStatus = MIXER_TRACKS_READY; 3266 } 3267 return mixerStatus; 3268} 3269 3270/* 3271The derived values that are cached: 3272 - mixBufferSize from frame count * frame size 3273 - activeSleepTime from activeSleepTimeUs() 3274 - idleSleepTime from idleSleepTimeUs() 3275 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3276 - maxPeriod from frame count and sample rate (MIXER only) 3277 3278The parameters that affect these derived values are: 3279 - frame count 3280 - frame size 3281 - sample rate 3282 - device type: A2DP or not 3283 - device latency 3284 - format: PCM or not 3285 - active sleep time 3286 - idle sleep time 3287*/ 3288 3289void AudioFlinger::PlaybackThread::cacheParameters_l() 3290{ 3291 mixBufferSize = mNormalFrameCount * mFrameSize; 3292 activeSleepTime = activeSleepTimeUs(); 3293 idleSleepTime = idleSleepTimeUs(); 3294} 3295 3296void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3297{ 3298 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3299 this, streamType, mTracks.size()); 3300 Mutex::Autolock _l(mLock); 3301 3302 size_t size = mTracks.size(); 3303 for (size_t i = 0; i < size; i++) { 3304 sp<Track> t = mTracks[i]; 3305 if (t->streamType() == streamType) { 3306 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3307 t->mCblk->cv.signal(); 3308 } 3309 } 3310} 3311 3312// getTrackName_l() must be called with ThreadBase::mLock held 3313int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3314{ 3315 return mAudioMixer->getTrackName(channelMask); 3316} 3317 3318// deleteTrackName_l() must be called with ThreadBase::mLock held 3319void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3320{ 3321 ALOGV("remove track (%d) and delete from mixer", name); 3322 mAudioMixer->deleteTrackName(name); 3323} 3324 3325// checkForNewParameters_l() must be called with ThreadBase::mLock held 3326bool AudioFlinger::MixerThread::checkForNewParameters_l() 3327{ 3328 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3329 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3330 bool reconfig = false; 3331 3332 while (!mNewParameters.isEmpty()) { 3333 3334 if (mFastMixer != NULL) { 3335 FastMixerStateQueue *sq = mFastMixer->sq(); 3336 FastMixerState *state = sq->begin(); 3337 if (!(state->mCommand & FastMixerState::IDLE)) { 3338 previousCommand = state->mCommand; 3339 state->mCommand = FastMixerState::HOT_IDLE; 3340 sq->end(); 3341 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3342 } else { 3343 sq->end(false /*didModify*/); 3344 } 3345 } 3346 3347 status_t status = NO_ERROR; 3348 String8 keyValuePair = mNewParameters[0]; 3349 AudioParameter param = AudioParameter(keyValuePair); 3350 int value; 3351 3352 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3353 reconfig = true; 3354 } 3355 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3356 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3357 status = BAD_VALUE; 3358 } else { 3359 reconfig = true; 3360 } 3361 } 3362 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3363 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3364 status = BAD_VALUE; 3365 } else { 3366 reconfig = true; 3367 } 3368 } 3369 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3370 // do not accept frame count changes if tracks are open as the track buffer 3371 // size depends on frame count and correct behavior would not be guaranteed 3372 // if frame count is changed after track creation 3373 if (!mTracks.isEmpty()) { 3374 status = INVALID_OPERATION; 3375 } else { 3376 reconfig = true; 3377 } 3378 } 3379 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3380#ifdef ADD_BATTERY_DATA 3381 // when changing the audio output device, call addBatteryData to notify 3382 // the change 3383 if ((int)mDevice != value) { 3384 uint32_t params = 0; 3385 // check whether speaker is on 3386 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3387 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3388 } 3389 3390 int deviceWithoutSpeaker 3391 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3392 // check if any other device (except speaker) is on 3393 if (value & deviceWithoutSpeaker ) { 3394 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3395 } 3396 3397 if (params != 0) { 3398 addBatteryData(params); 3399 } 3400 } 3401#endif 3402 3403 // forward device change to effects that have requested to be 3404 // aware of attached audio device. 3405 mDevice = (uint32_t)value; 3406 for (size_t i = 0; i < mEffectChains.size(); i++) { 3407 mEffectChains[i]->setDevice_l(mDevice); 3408 } 3409 } 3410 3411 if (status == NO_ERROR) { 3412 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3413 keyValuePair.string()); 3414 if (!mStandby && status == INVALID_OPERATION) { 3415 mOutput->stream->common.standby(&mOutput->stream->common); 3416 mStandby = true; 3417 mBytesWritten = 0; 3418 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3419 keyValuePair.string()); 3420 } 3421 if (status == NO_ERROR && reconfig) { 3422 delete mAudioMixer; 3423 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3424 mAudioMixer = NULL; 3425 readOutputParameters(); 3426 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3427 for (size_t i = 0; i < mTracks.size() ; i++) { 3428 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3429 if (name < 0) break; 3430 mTracks[i]->mName = name; 3431 // limit track sample rate to 2 x new output sample rate 3432 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3433 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3434 } 3435 } 3436 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3437 } 3438 } 3439 3440 mNewParameters.removeAt(0); 3441 3442 mParamStatus = status; 3443 mParamCond.signal(); 3444 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3445 // already timed out waiting for the status and will never signal the condition. 3446 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3447 } 3448 3449 if (!(previousCommand & FastMixerState::IDLE)) { 3450 ALOG_ASSERT(mFastMixer != NULL); 3451 FastMixerStateQueue *sq = mFastMixer->sq(); 3452 FastMixerState *state = sq->begin(); 3453 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3454 state->mCommand = previousCommand; 3455 sq->end(); 3456 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3457 } 3458 3459 return reconfig; 3460} 3461 3462status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3463{ 3464 const size_t SIZE = 256; 3465 char buffer[SIZE]; 3466 String8 result; 3467 3468 PlaybackThread::dumpInternals(fd, args); 3469 3470 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3471 result.append(buffer); 3472 write(fd, result.string(), result.size()); 3473 3474 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3475 FastMixerDumpState copy = mFastMixerDumpState; 3476 copy.dump(fd); 3477 3478 // Write the tee output to a .wav file 3479 NBAIO_Source *teeSource = mTeeSource.get(); 3480 if (teeSource != NULL) { 3481 char teePath[64]; 3482 struct timeval tv; 3483 gettimeofday(&tv, NULL); 3484 struct tm tm; 3485 localtime_r(&tv.tv_sec, &tm); 3486 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3487 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3488 if (teeFd >= 0) { 3489 char wavHeader[44]; 3490 memcpy(wavHeader, 3491 "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", 3492 sizeof(wavHeader)); 3493 NBAIO_Format format = teeSource->format(); 3494 unsigned channelCount = Format_channelCount(format); 3495 ALOG_ASSERT(channelCount <= FCC_2); 3496 unsigned sampleRate = Format_sampleRate(format); 3497 wavHeader[22] = channelCount; // number of channels 3498 wavHeader[24] = sampleRate; // sample rate 3499 wavHeader[25] = sampleRate >> 8; 3500 wavHeader[32] = channelCount * 2; // block alignment 3501 write(teeFd, wavHeader, sizeof(wavHeader)); 3502 size_t total = 0; 3503 bool firstRead = true; 3504 for (;;) { 3505#define TEE_SINK_READ 1024 3506 short buffer[TEE_SINK_READ * FCC_2]; 3507 size_t count = TEE_SINK_READ; 3508 ssize_t actual = teeSource->read(buffer, count); 3509 bool wasFirstRead = firstRead; 3510 firstRead = false; 3511 if (actual <= 0) { 3512 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3513 continue; 3514 } 3515 break; 3516 } 3517 ALOG_ASSERT(actual <= count); 3518 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3519 total += actual; 3520 } 3521 lseek(teeFd, (off_t) 4, SEEK_SET); 3522 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3523 write(teeFd, &temp, sizeof(temp)); 3524 lseek(teeFd, (off_t) 40, SEEK_SET); 3525 temp = total * channelCount * sizeof(short); 3526 write(teeFd, &temp, sizeof(temp)); 3527 close(teeFd); 3528 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3529 } else { 3530 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3531 } 3532 } 3533 3534 return NO_ERROR; 3535} 3536 3537uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3538{ 3539 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3540} 3541 3542uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3543{ 3544 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3545} 3546 3547void AudioFlinger::MixerThread::cacheParameters_l() 3548{ 3549 PlaybackThread::cacheParameters_l(); 3550 3551 // FIXME: Relaxed timing because of a certain device that can't meet latency 3552 // Should be reduced to 2x after the vendor fixes the driver issue 3553 // increase threshold again due to low power audio mode. The way this warning 3554 // threshold is calculated and its usefulness should be reconsidered anyway. 3555 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3556} 3557 3558// ---------------------------------------------------------------------------- 3559AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3560 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3561 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3562 // mLeftVolFloat, mRightVolFloat 3563 // mLeftVolShort, mRightVolShort 3564{ 3565} 3566 3567AudioFlinger::DirectOutputThread::~DirectOutputThread() 3568{ 3569} 3570 3571AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3572 Vector< sp<Track> > *tracksToRemove 3573) 3574{ 3575 sp<Track> trackToRemove; 3576 3577 mixer_state mixerStatus = MIXER_IDLE; 3578 3579 // find out which tracks need to be processed 3580 if (mActiveTracks.size() != 0) { 3581 sp<Track> t = mActiveTracks[0].promote(); 3582 // The track died recently 3583 if (t == 0) return MIXER_IDLE; 3584 3585 Track* const track = t.get(); 3586 audio_track_cblk_t* cblk = track->cblk(); 3587 3588 // The first time a track is added we wait 3589 // for all its buffers to be filled before processing it 3590 if (cblk->framesReady() && track->isReady() && 3591 !track->isPaused() && !track->isTerminated()) 3592 { 3593 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3594 3595 if (track->mFillingUpStatus == Track::FS_FILLED) { 3596 track->mFillingUpStatus = Track::FS_ACTIVE; 3597 mLeftVolFloat = mRightVolFloat = 0; 3598 mLeftVolShort = mRightVolShort = 0; 3599 if (track->mState == TrackBase::RESUMING) { 3600 track->mState = TrackBase::ACTIVE; 3601 rampVolume = true; 3602 } 3603 } else if (cblk->server != 0) { 3604 // If the track is stopped before the first frame was mixed, 3605 // do not apply ramp 3606 rampVolume = true; 3607 } 3608 // compute volume for this track 3609 float left, right; 3610 if (track->isMuted() || mMasterMute || track->isPausing() || 3611 mStreamTypes[track->streamType()].mute) { 3612 left = right = 0; 3613 if (track->isPausing()) { 3614 track->setPaused(); 3615 } 3616 } else { 3617 float typeVolume = mStreamTypes[track->streamType()].volume; 3618 float v = mMasterVolume * typeVolume; 3619 uint32_t vlr = cblk->getVolumeLR(); 3620 float v_clamped = v * (vlr & 0xFFFF); 3621 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3622 left = v_clamped/MAX_GAIN; 3623 v_clamped = v * (vlr >> 16); 3624 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3625 right = v_clamped/MAX_GAIN; 3626 } 3627 3628 if (left != mLeftVolFloat || right != mRightVolFloat) { 3629 mLeftVolFloat = left; 3630 mRightVolFloat = right; 3631 3632 // If audio HAL implements volume control, 3633 // force software volume to nominal value 3634 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3635 left = 1.0f; 3636 right = 1.0f; 3637 } 3638 3639 // Convert volumes from float to 8.24 3640 uint32_t vl = (uint32_t)(left * (1 << 24)); 3641 uint32_t vr = (uint32_t)(right * (1 << 24)); 3642 3643 // Delegate volume control to effect in track effect chain if needed 3644 // only one effect chain can be present on DirectOutputThread, so if 3645 // there is one, the track is connected to it 3646 if (!mEffectChains.isEmpty()) { 3647 // Do not ramp volume if volume is controlled by effect 3648 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3649 rampVolume = false; 3650 } 3651 } 3652 3653 // Convert volumes from 8.24 to 4.12 format 3654 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3655 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3656 leftVol = (uint16_t)v_clamped; 3657 v_clamped = (vr + (1 << 11)) >> 12; 3658 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3659 rightVol = (uint16_t)v_clamped; 3660 } else { 3661 leftVol = mLeftVolShort; 3662 rightVol = mRightVolShort; 3663 rampVolume = false; 3664 } 3665 3666 // reset retry count 3667 track->mRetryCount = kMaxTrackRetriesDirect; 3668 mActiveTrack = t; 3669 mixerStatus = MIXER_TRACKS_READY; 3670 } else { 3671 // clear effect chain input buffer if an active track underruns to avoid sending 3672 // previous audio buffer again to effects 3673 if (!mEffectChains.isEmpty()) { 3674 mEffectChains[0]->clearInputBuffer(); 3675 } 3676 3677 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3678 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3679 // We have consumed all the buffers of this track. 3680 // Remove it from the list of active tracks. 3681 // TODO: implement behavior for compressed audio 3682 size_t audioHALFrames = 3683 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3684 size_t framesWritten = 3685 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3686 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3687 if (track->isStopped()) { 3688 track->reset(); 3689 } 3690 trackToRemove = track; 3691 } 3692 } else { 3693 // No buffers for this track. Give it a few chances to 3694 // fill a buffer, then remove it from active list. 3695 if (--(track->mRetryCount) <= 0) { 3696 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3697 trackToRemove = track; 3698 } else { 3699 mixerStatus = MIXER_TRACKS_ENABLED; 3700 } 3701 } 3702 } 3703 } 3704 3705 // FIXME merge this with similar code for removing multiple tracks 3706 // remove all the tracks that need to be... 3707 if (CC_UNLIKELY(trackToRemove != 0)) { 3708 tracksToRemove->add(trackToRemove); 3709 mActiveTracks.remove(trackToRemove); 3710 if (!mEffectChains.isEmpty()) { 3711 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3712 trackToRemove->sessionId()); 3713 mEffectChains[0]->decActiveTrackCnt(); 3714 } 3715 if (trackToRemove->isTerminated()) { 3716 removeTrack_l(trackToRemove); 3717 } 3718 } 3719 3720 return mixerStatus; 3721} 3722 3723void AudioFlinger::DirectOutputThread::threadLoop_mix() 3724{ 3725 AudioBufferProvider::Buffer buffer; 3726 size_t frameCount = mFrameCount; 3727 int8_t *curBuf = (int8_t *)mMixBuffer; 3728 // output audio to hardware 3729 while (frameCount) { 3730 buffer.frameCount = frameCount; 3731 mActiveTrack->getNextBuffer(&buffer); 3732 if (CC_UNLIKELY(buffer.raw == NULL)) { 3733 memset(curBuf, 0, frameCount * mFrameSize); 3734 break; 3735 } 3736 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3737 frameCount -= buffer.frameCount; 3738 curBuf += buffer.frameCount * mFrameSize; 3739 mActiveTrack->releaseBuffer(&buffer); 3740 } 3741 sleepTime = 0; 3742 standbyTime = systemTime() + standbyDelay; 3743 mActiveTrack.clear(); 3744 3745 // apply volume 3746 3747 // Do not apply volume on compressed audio 3748 if (!audio_is_linear_pcm(mFormat)) { 3749 return; 3750 } 3751 3752 // convert to signed 16 bit before volume calculation 3753 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3754 size_t count = mFrameCount * mChannelCount; 3755 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3756 int16_t *dst = mMixBuffer + count-1; 3757 while (count--) { 3758 *dst-- = (int16_t)(*src--^0x80) << 8; 3759 } 3760 } 3761 3762 frameCount = mFrameCount; 3763 int16_t *out = mMixBuffer; 3764 if (rampVolume) { 3765 if (mChannelCount == 1) { 3766 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3767 int32_t vlInc = d / (int32_t)frameCount; 3768 int32_t vl = ((int32_t)mLeftVolShort << 16); 3769 do { 3770 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3771 out++; 3772 vl += vlInc; 3773 } while (--frameCount); 3774 3775 } else { 3776 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3777 int32_t vlInc = d / (int32_t)frameCount; 3778 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3779 int32_t vrInc = d / (int32_t)frameCount; 3780 int32_t vl = ((int32_t)mLeftVolShort << 16); 3781 int32_t vr = ((int32_t)mRightVolShort << 16); 3782 do { 3783 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3784 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3785 out += 2; 3786 vl += vlInc; 3787 vr += vrInc; 3788 } while (--frameCount); 3789 } 3790 } else { 3791 if (mChannelCount == 1) { 3792 do { 3793 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3794 out++; 3795 } while (--frameCount); 3796 } else { 3797 do { 3798 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3799 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3800 out += 2; 3801 } while (--frameCount); 3802 } 3803 } 3804 3805 // convert back to unsigned 8 bit after volume calculation 3806 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3807 size_t count = mFrameCount * mChannelCount; 3808 int16_t *src = mMixBuffer; 3809 uint8_t *dst = (uint8_t *)mMixBuffer; 3810 while (count--) { 3811 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3812 } 3813 } 3814 3815 mLeftVolShort = leftVol; 3816 mRightVolShort = rightVol; 3817} 3818 3819void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3820{ 3821 if (sleepTime == 0) { 3822 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3823 sleepTime = activeSleepTime; 3824 } else { 3825 sleepTime = idleSleepTime; 3826 } 3827 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3828 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3829 sleepTime = 0; 3830 } 3831} 3832 3833// getTrackName_l() must be called with ThreadBase::mLock held 3834int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3835{ 3836 return 0; 3837} 3838 3839// deleteTrackName_l() must be called with ThreadBase::mLock held 3840void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3841{ 3842} 3843 3844// checkForNewParameters_l() must be called with ThreadBase::mLock held 3845bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3846{ 3847 bool reconfig = false; 3848 3849 while (!mNewParameters.isEmpty()) { 3850 status_t status = NO_ERROR; 3851 String8 keyValuePair = mNewParameters[0]; 3852 AudioParameter param = AudioParameter(keyValuePair); 3853 int value; 3854 3855 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3856 // do not accept frame count changes if tracks are open as the track buffer 3857 // size depends on frame count and correct behavior would not be garantied 3858 // if frame count is changed after track creation 3859 if (!mTracks.isEmpty()) { 3860 status = INVALID_OPERATION; 3861 } else { 3862 reconfig = true; 3863 } 3864 } 3865 if (status == NO_ERROR) { 3866 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3867 keyValuePair.string()); 3868 if (!mStandby && status == INVALID_OPERATION) { 3869 mOutput->stream->common.standby(&mOutput->stream->common); 3870 mStandby = true; 3871 mBytesWritten = 0; 3872 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3873 keyValuePair.string()); 3874 } 3875 if (status == NO_ERROR && reconfig) { 3876 readOutputParameters(); 3877 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3878 } 3879 } 3880 3881 mNewParameters.removeAt(0); 3882 3883 mParamStatus = status; 3884 mParamCond.signal(); 3885 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3886 // already timed out waiting for the status and will never signal the condition. 3887 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3888 } 3889 return reconfig; 3890} 3891 3892uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3893{ 3894 uint32_t time; 3895 if (audio_is_linear_pcm(mFormat)) { 3896 time = PlaybackThread::activeSleepTimeUs(); 3897 } else { 3898 time = 10000; 3899 } 3900 return time; 3901} 3902 3903uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3904{ 3905 uint32_t time; 3906 if (audio_is_linear_pcm(mFormat)) { 3907 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3908 } else { 3909 time = 10000; 3910 } 3911 return time; 3912} 3913 3914uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3915{ 3916 uint32_t time; 3917 if (audio_is_linear_pcm(mFormat)) { 3918 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3919 } else { 3920 time = 10000; 3921 } 3922 return time; 3923} 3924 3925void AudioFlinger::DirectOutputThread::cacheParameters_l() 3926{ 3927 PlaybackThread::cacheParameters_l(); 3928 3929 // use shorter standby delay as on normal output to release 3930 // hardware resources as soon as possible 3931 standbyDelay = microseconds(activeSleepTime*2); 3932} 3933 3934// ---------------------------------------------------------------------------- 3935 3936AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3937 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3938 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3939 mWaitTimeMs(UINT_MAX) 3940{ 3941 addOutputTrack(mainThread); 3942} 3943 3944AudioFlinger::DuplicatingThread::~DuplicatingThread() 3945{ 3946 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3947 mOutputTracks[i]->destroy(); 3948 } 3949} 3950 3951void AudioFlinger::DuplicatingThread::threadLoop_mix() 3952{ 3953 // mix buffers... 3954 if (outputsReady(outputTracks)) { 3955 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3956 } else { 3957 memset(mMixBuffer, 0, mixBufferSize); 3958 } 3959 sleepTime = 0; 3960 writeFrames = mNormalFrameCount; 3961} 3962 3963void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3964{ 3965 if (sleepTime == 0) { 3966 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3967 sleepTime = activeSleepTime; 3968 } else { 3969 sleepTime = idleSleepTime; 3970 } 3971 } else if (mBytesWritten != 0) { 3972 // flush remaining overflow buffers in output tracks 3973 for (size_t i = 0; i < outputTracks.size(); i++) { 3974 if (outputTracks[i]->isActive()) { 3975 sleepTime = 0; 3976 writeFrames = 0; 3977 memset(mMixBuffer, 0, mixBufferSize); 3978 break; 3979 } 3980 } 3981 } 3982} 3983 3984void AudioFlinger::DuplicatingThread::threadLoop_write() 3985{ 3986 standbyTime = systemTime() + standbyDelay; 3987 for (size_t i = 0; i < outputTracks.size(); i++) { 3988 outputTracks[i]->write(mMixBuffer, writeFrames); 3989 } 3990 mBytesWritten += mixBufferSize; 3991} 3992 3993void AudioFlinger::DuplicatingThread::threadLoop_standby() 3994{ 3995 // DuplicatingThread implements standby by stopping all tracks 3996 for (size_t i = 0; i < outputTracks.size(); i++) { 3997 outputTracks[i]->stop(); 3998 } 3999} 4000 4001void AudioFlinger::DuplicatingThread::saveOutputTracks() 4002{ 4003 outputTracks = mOutputTracks; 4004} 4005 4006void AudioFlinger::DuplicatingThread::clearOutputTracks() 4007{ 4008 outputTracks.clear(); 4009} 4010 4011void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4012{ 4013 Mutex::Autolock _l(mLock); 4014 // FIXME explain this formula 4015 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4016 OutputTrack *outputTrack = new OutputTrack(thread, 4017 this, 4018 mSampleRate, 4019 mFormat, 4020 mChannelMask, 4021 frameCount); 4022 if (outputTrack->cblk() != NULL) { 4023 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4024 mOutputTracks.add(outputTrack); 4025 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4026 updateWaitTime_l(); 4027 } 4028} 4029 4030void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4031{ 4032 Mutex::Autolock _l(mLock); 4033 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4034 if (mOutputTracks[i]->thread() == thread) { 4035 mOutputTracks[i]->destroy(); 4036 mOutputTracks.removeAt(i); 4037 updateWaitTime_l(); 4038 return; 4039 } 4040 } 4041 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4042} 4043 4044// caller must hold mLock 4045void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4046{ 4047 mWaitTimeMs = UINT_MAX; 4048 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4049 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4050 if (strong != 0) { 4051 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4052 if (waitTimeMs < mWaitTimeMs) { 4053 mWaitTimeMs = waitTimeMs; 4054 } 4055 } 4056 } 4057} 4058 4059 4060bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4061{ 4062 for (size_t i = 0; i < outputTracks.size(); i++) { 4063 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4064 if (thread == 0) { 4065 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4066 return false; 4067 } 4068 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4069 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4070 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4071 return false; 4072 } 4073 } 4074 return true; 4075} 4076 4077uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4078{ 4079 return (mWaitTimeMs * 1000) / 2; 4080} 4081 4082void AudioFlinger::DuplicatingThread::cacheParameters_l() 4083{ 4084 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4085 updateWaitTime_l(); 4086 4087 MixerThread::cacheParameters_l(); 4088} 4089 4090// ---------------------------------------------------------------------------- 4091 4092// TrackBase constructor must be called with AudioFlinger::mLock held 4093AudioFlinger::ThreadBase::TrackBase::TrackBase( 4094 ThreadBase *thread, 4095 const sp<Client>& client, 4096 uint32_t sampleRate, 4097 audio_format_t format, 4098 uint32_t channelMask, 4099 int frameCount, 4100 const sp<IMemory>& sharedBuffer, 4101 int sessionId) 4102 : RefBase(), 4103 mThread(thread), 4104 mClient(client), 4105 mCblk(NULL), 4106 // mBuffer 4107 // mBufferEnd 4108 mFrameCount(0), 4109 mState(IDLE), 4110 mSampleRate(sampleRate), 4111 mFormat(format), 4112 mStepServerFailed(false), 4113 mSessionId(sessionId) 4114 // mChannelCount 4115 // mChannelMask 4116{ 4117 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4118 4119 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4120 size_t size = sizeof(audio_track_cblk_t); 4121 uint8_t channelCount = popcount(channelMask); 4122 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4123 if (sharedBuffer == 0) { 4124 size += bufferSize; 4125 } 4126 4127 if (client != NULL) { 4128 mCblkMemory = client->heap()->allocate(size); 4129 if (mCblkMemory != 0) { 4130 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4131 if (mCblk != NULL) { // construct the shared structure in-place. 4132 new(mCblk) audio_track_cblk_t(); 4133 // clear all buffers 4134 mCblk->frameCount = frameCount; 4135 mCblk->sampleRate = sampleRate; 4136// uncomment the following lines to quickly test 32-bit wraparound 4137// mCblk->user = 0xffff0000; 4138// mCblk->server = 0xffff0000; 4139// mCblk->userBase = 0xffff0000; 4140// mCblk->serverBase = 0xffff0000; 4141 mChannelCount = channelCount; 4142 mChannelMask = channelMask; 4143 if (sharedBuffer == 0) { 4144 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4145 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4146 // Force underrun condition to avoid false underrun callback until first data is 4147 // written to buffer (other flags are cleared) 4148 mCblk->flags = CBLK_UNDERRUN_ON; 4149 } else { 4150 mBuffer = sharedBuffer->pointer(); 4151 } 4152 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4153 } 4154 } else { 4155 ALOGE("not enough memory for AudioTrack size=%u", size); 4156 client->heap()->dump("AudioTrack"); 4157 return; 4158 } 4159 } else { 4160 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4161 // construct the shared structure in-place. 4162 new(mCblk) audio_track_cblk_t(); 4163 // clear all buffers 4164 mCblk->frameCount = frameCount; 4165 mCblk->sampleRate = sampleRate; 4166// uncomment the following lines to quickly test 32-bit wraparound 4167// mCblk->user = 0xffff0000; 4168// mCblk->server = 0xffff0000; 4169// mCblk->userBase = 0xffff0000; 4170// mCblk->serverBase = 0xffff0000; 4171 mChannelCount = channelCount; 4172 mChannelMask = channelMask; 4173 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4174 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4175 // Force underrun condition to avoid false underrun callback until first data is 4176 // written to buffer (other flags are cleared) 4177 mCblk->flags = CBLK_UNDERRUN_ON; 4178 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4179 } 4180} 4181 4182AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4183{ 4184 if (mCblk != NULL) { 4185 if (mClient == 0) { 4186 delete mCblk; 4187 } else { 4188 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4189 } 4190 } 4191 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4192 if (mClient != 0) { 4193 // Client destructor must run with AudioFlinger mutex locked 4194 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4195 // If the client's reference count drops to zero, the associated destructor 4196 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4197 // relying on the automatic clear() at end of scope. 4198 mClient.clear(); 4199 } 4200} 4201 4202// AudioBufferProvider interface 4203// getNextBuffer() = 0; 4204// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4205void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4206{ 4207 buffer->raw = NULL; 4208 mFrameCount = buffer->frameCount; 4209 // FIXME See note at getNextBuffer() 4210 (void) step(); // ignore return value of step() 4211 buffer->frameCount = 0; 4212} 4213 4214bool AudioFlinger::ThreadBase::TrackBase::step() { 4215 bool result; 4216 audio_track_cblk_t* cblk = this->cblk(); 4217 4218 result = cblk->stepServer(mFrameCount); 4219 if (!result) { 4220 ALOGV("stepServer failed acquiring cblk mutex"); 4221 mStepServerFailed = true; 4222 } 4223 return result; 4224} 4225 4226void AudioFlinger::ThreadBase::TrackBase::reset() { 4227 audio_track_cblk_t* cblk = this->cblk(); 4228 4229 cblk->user = 0; 4230 cblk->server = 0; 4231 cblk->userBase = 0; 4232 cblk->serverBase = 0; 4233 mStepServerFailed = false; 4234 ALOGV("TrackBase::reset"); 4235} 4236 4237int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4238 return (int)mCblk->sampleRate; 4239} 4240 4241void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4242 audio_track_cblk_t* cblk = this->cblk(); 4243 size_t frameSize = cblk->frameSize; 4244 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4245 int8_t *bufferEnd = bufferStart + frames * frameSize; 4246 4247 // Check validity of returned pointer in case the track control block would have been corrupted. 4248 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4249 "TrackBase::getBuffer buffer out of range:\n" 4250 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4251 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4252 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4253 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4254 4255 return bufferStart; 4256} 4257 4258status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4259{ 4260 mSyncEvents.add(event); 4261 return NO_ERROR; 4262} 4263 4264// ---------------------------------------------------------------------------- 4265 4266// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4267AudioFlinger::PlaybackThread::Track::Track( 4268 PlaybackThread *thread, 4269 const sp<Client>& client, 4270 audio_stream_type_t streamType, 4271 uint32_t sampleRate, 4272 audio_format_t format, 4273 uint32_t channelMask, 4274 int frameCount, 4275 const sp<IMemory>& sharedBuffer, 4276 int sessionId, 4277 IAudioFlinger::track_flags_t flags) 4278 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4279 mMute(false), 4280 mFillingUpStatus(FS_INVALID), 4281 // mRetryCount initialized later when needed 4282 mSharedBuffer(sharedBuffer), 4283 mStreamType(streamType), 4284 mName(-1), // see note below 4285 mMainBuffer(thread->mixBuffer()), 4286 mAuxBuffer(NULL), 4287 mAuxEffectId(0), mHasVolumeController(false), 4288 mPresentationCompleteFrames(0), 4289 mFlags(flags), 4290 mFastIndex(-1), 4291 mUnderrunCount(0), 4292 mCachedVolume(1.0) 4293{ 4294 if (mCblk != NULL) { 4295 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4296 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4297 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4298 if (flags & IAudioFlinger::TRACK_FAST) { 4299 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4300 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4301 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4302 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4303 // FIXME This is too eager. We allocate a fast track index before the 4304 // fast track becomes active. Since fast tracks are a scarce resource, 4305 // this means we are potentially denying other more important fast tracks from 4306 // being created. It would be better to allocate the index dynamically. 4307 mFastIndex = i; 4308 // Read the initial underruns because this field is never cleared by the fast mixer 4309 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4310 thread->mFastTrackAvailMask &= ~(1 << i); 4311 } 4312 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4313 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4314 if (mName < 0) { 4315 ALOGE("no more track names available"); 4316 // FIXME bug - if sufficient fast track indices, but insufficient normal mixer names, 4317 // then we leak a fast track index. Should swap these two sections, or better yet 4318 // only allocate a normal mixer name for normal tracks. 4319 } 4320 } 4321 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4322} 4323 4324AudioFlinger::PlaybackThread::Track::~Track() 4325{ 4326 ALOGV("PlaybackThread::Track destructor"); 4327 sp<ThreadBase> thread = mThread.promote(); 4328 if (thread != 0) { 4329 Mutex::Autolock _l(thread->mLock); 4330 mState = TERMINATED; 4331 } 4332} 4333 4334void AudioFlinger::PlaybackThread::Track::destroy() 4335{ 4336 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4337 // by removing it from mTracks vector, so there is a risk that this Tracks's 4338 // destructor is called. As the destructor needs to lock mLock, 4339 // we must acquire a strong reference on this Track before locking mLock 4340 // here so that the destructor is called only when exiting this function. 4341 // On the other hand, as long as Track::destroy() is only called by 4342 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4343 // this Track with its member mTrack. 4344 sp<Track> keep(this); 4345 { // scope for mLock 4346 sp<ThreadBase> thread = mThread.promote(); 4347 if (thread != 0) { 4348 if (!isOutputTrack()) { 4349 if (mState == ACTIVE || mState == RESUMING) { 4350 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4351 4352#ifdef ADD_BATTERY_DATA 4353 // to track the speaker usage 4354 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4355#endif 4356 } 4357 AudioSystem::releaseOutput(thread->id()); 4358 } 4359 Mutex::Autolock _l(thread->mLock); 4360 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4361 playbackThread->destroyTrack_l(this); 4362 } 4363 } 4364} 4365 4366/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4367{ 4368 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4369 " Server User Main buf Aux Buf Flags FastUnder\n"); 4370} 4371 4372void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4373{ 4374 uint32_t vlr = mCblk->getVolumeLR(); 4375 if (isFastTrack()) { 4376 sprintf(buffer, " F %2d", mFastIndex); 4377 } else { 4378 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4379 } 4380 track_state state = mState; 4381 char stateChar; 4382 switch (state) { 4383 case IDLE: 4384 stateChar = 'I'; 4385 break; 4386 case TERMINATED: 4387 stateChar = 'T'; 4388 break; 4389 case STOPPING_1: 4390 stateChar = 's'; 4391 break; 4392 case STOPPING_2: 4393 stateChar = '5'; 4394 break; 4395 case STOPPED: 4396 stateChar = 'S'; 4397 break; 4398 case RESUMING: 4399 stateChar = 'R'; 4400 break; 4401 case ACTIVE: 4402 stateChar = 'A'; 4403 break; 4404 case PAUSING: 4405 stateChar = 'p'; 4406 break; 4407 case PAUSED: 4408 stateChar = 'P'; 4409 break; 4410 case FLUSHED: 4411 stateChar = 'F'; 4412 break; 4413 default: 4414 stateChar = '?'; 4415 break; 4416 } 4417 char nowInUnderrun; 4418 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4419 case UNDERRUN_FULL: 4420 nowInUnderrun = ' '; 4421 break; 4422 case UNDERRUN_PARTIAL: 4423 nowInUnderrun = '<'; 4424 break; 4425 case UNDERRUN_EMPTY: 4426 nowInUnderrun = '*'; 4427 break; 4428 default: 4429 nowInUnderrun = '?'; 4430 break; 4431 } 4432 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4433 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4434 (mClient == 0) ? getpid_cached : mClient->pid(), 4435 mStreamType, 4436 mFormat, 4437 mChannelMask, 4438 mSessionId, 4439 mFrameCount, 4440 mCblk->frameCount, 4441 stateChar, 4442 mMute, 4443 mFillingUpStatus, 4444 mCblk->sampleRate, 4445 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4446 20.0 * log10((vlr >> 16) / 4096.0), 4447 mCblk->server, 4448 mCblk->user, 4449 (int)mMainBuffer, 4450 (int)mAuxBuffer, 4451 mCblk->flags, 4452 mUnderrunCount, 4453 nowInUnderrun); 4454} 4455 4456// AudioBufferProvider interface 4457status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4458 AudioBufferProvider::Buffer* buffer, int64_t pts) 4459{ 4460 audio_track_cblk_t* cblk = this->cblk(); 4461 uint32_t framesReady; 4462 uint32_t framesReq = buffer->frameCount; 4463 4464 // Check if last stepServer failed, try to step now 4465 if (mStepServerFailed) { 4466 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4467 // Since the fast mixer is higher priority than client callback thread, 4468 // it does not result in priority inversion for client. 4469 // But a non-blocking solution would be preferable to avoid 4470 // fast mixer being unable to tryLock(), and 4471 // to avoid the extra context switches if the client wakes up, 4472 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4473 if (!step()) goto getNextBuffer_exit; 4474 ALOGV("stepServer recovered"); 4475 mStepServerFailed = false; 4476 } 4477 4478 // FIXME Same as above 4479 framesReady = cblk->framesReady(); 4480 4481 if (CC_LIKELY(framesReady)) { 4482 uint32_t s = cblk->server; 4483 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4484 4485 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4486 if (framesReq > framesReady) { 4487 framesReq = framesReady; 4488 } 4489 if (framesReq > bufferEnd - s) { 4490 framesReq = bufferEnd - s; 4491 } 4492 4493 buffer->raw = getBuffer(s, framesReq); 4494 if (buffer->raw == NULL) goto getNextBuffer_exit; 4495 4496 buffer->frameCount = framesReq; 4497 return NO_ERROR; 4498 } 4499 4500getNextBuffer_exit: 4501 buffer->raw = NULL; 4502 buffer->frameCount = 0; 4503 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4504 return NOT_ENOUGH_DATA; 4505} 4506 4507// Note that framesReady() takes a mutex on the control block using tryLock(). 4508// This could result in priority inversion if framesReady() is called by the normal mixer, 4509// as the normal mixer thread runs at lower 4510// priority than the client's callback thread: there is a short window within framesReady() 4511// during which the normal mixer could be preempted, and the client callback would block. 4512// Another problem can occur if framesReady() is called by the fast mixer: 4513// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4514// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4515size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4516 return mCblk->framesReady(); 4517} 4518 4519// Don't call for fast tracks; the framesReady() could result in priority inversion 4520bool AudioFlinger::PlaybackThread::Track::isReady() const { 4521 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4522 4523 if (framesReady() >= mCblk->frameCount || 4524 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4525 mFillingUpStatus = FS_FILLED; 4526 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4527 return true; 4528 } 4529 return false; 4530} 4531 4532status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4533 int triggerSession) 4534{ 4535 status_t status = NO_ERROR; 4536 ALOGV("start(%d), calling pid %d session %d", 4537 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4538 4539 sp<ThreadBase> thread = mThread.promote(); 4540 if (thread != 0) { 4541 Mutex::Autolock _l(thread->mLock); 4542 track_state state = mState; 4543 // here the track could be either new, or restarted 4544 // in both cases "unstop" the track 4545 if (mState == PAUSED) { 4546 mState = TrackBase::RESUMING; 4547 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4548 } else { 4549 mState = TrackBase::ACTIVE; 4550 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4551 } 4552 4553 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4554 thread->mLock.unlock(); 4555 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4556 thread->mLock.lock(); 4557 4558#ifdef ADD_BATTERY_DATA 4559 // to track the speaker usage 4560 if (status == NO_ERROR) { 4561 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4562 } 4563#endif 4564 } 4565 if (status == NO_ERROR) { 4566 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4567 playbackThread->addTrack_l(this); 4568 } else { 4569 mState = state; 4570 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4571 } 4572 } else { 4573 status = BAD_VALUE; 4574 } 4575 return status; 4576} 4577 4578void AudioFlinger::PlaybackThread::Track::stop() 4579{ 4580 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4581 sp<ThreadBase> thread = mThread.promote(); 4582 if (thread != 0) { 4583 Mutex::Autolock _l(thread->mLock); 4584 track_state state = mState; 4585 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4586 // If the track is not active (PAUSED and buffers full), flush buffers 4587 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4588 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4589 reset(); 4590 mState = STOPPED; 4591 } else if (!isFastTrack()) { 4592 mState = STOPPED; 4593 } else { 4594 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4595 // and then to STOPPED and reset() when presentation is complete 4596 mState = STOPPING_1; 4597 } 4598 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4599 } 4600 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4601 thread->mLock.unlock(); 4602 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4603 thread->mLock.lock(); 4604 4605#ifdef ADD_BATTERY_DATA 4606 // to track the speaker usage 4607 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4608#endif 4609 } 4610 } 4611} 4612 4613void AudioFlinger::PlaybackThread::Track::pause() 4614{ 4615 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4616 sp<ThreadBase> thread = mThread.promote(); 4617 if (thread != 0) { 4618 Mutex::Autolock _l(thread->mLock); 4619 if (mState == ACTIVE || mState == RESUMING) { 4620 mState = PAUSING; 4621 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4622 if (!isOutputTrack()) { 4623 thread->mLock.unlock(); 4624 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4625 thread->mLock.lock(); 4626 4627#ifdef ADD_BATTERY_DATA 4628 // to track the speaker usage 4629 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4630#endif 4631 } 4632 } 4633 } 4634} 4635 4636void AudioFlinger::PlaybackThread::Track::flush() 4637{ 4638 ALOGV("flush(%d)", mName); 4639 sp<ThreadBase> thread = mThread.promote(); 4640 if (thread != 0) { 4641 Mutex::Autolock _l(thread->mLock); 4642 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4643 mState != PAUSING) { 4644 return; 4645 } 4646 // No point remaining in PAUSED state after a flush => go to 4647 // FLUSHED state 4648 mState = FLUSHED; 4649 // do not reset the track if it is still in the process of being stopped or paused. 4650 // this will be done by prepareTracks_l() when the track is stopped. 4651 // prepareTracks_l() will see mState == FLUSHED, then 4652 // remove from active track list, reset(), and trigger presentation complete 4653 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4654 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4655 reset(); 4656 } 4657 } 4658} 4659 4660void AudioFlinger::PlaybackThread::Track::reset() 4661{ 4662 // Do not reset twice to avoid discarding data written just after a flush and before 4663 // the audioflinger thread detects the track is stopped. 4664 if (!mResetDone) { 4665 TrackBase::reset(); 4666 // Force underrun condition to avoid false underrun callback until first data is 4667 // written to buffer 4668 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4669 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4670 mFillingUpStatus = FS_FILLING; 4671 mResetDone = true; 4672 if (mState == FLUSHED) { 4673 mState = IDLE; 4674 } 4675 } 4676} 4677 4678void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4679{ 4680 mMute = muted; 4681} 4682 4683status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4684{ 4685 status_t status = DEAD_OBJECT; 4686 sp<ThreadBase> thread = mThread.promote(); 4687 if (thread != 0) { 4688 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4689 status = playbackThread->attachAuxEffect(this, EffectId); 4690 } 4691 return status; 4692} 4693 4694void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4695{ 4696 mAuxEffectId = EffectId; 4697 mAuxBuffer = buffer; 4698} 4699 4700bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4701 size_t audioHalFrames) 4702{ 4703 // a track is considered presented when the total number of frames written to audio HAL 4704 // corresponds to the number of frames written when presentationComplete() is called for the 4705 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4706 if (mPresentationCompleteFrames == 0) { 4707 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4708 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4709 mPresentationCompleteFrames, audioHalFrames); 4710 } 4711 if (framesWritten >= mPresentationCompleteFrames) { 4712 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4713 mSessionId, framesWritten); 4714 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4715 return true; 4716 } 4717 return false; 4718} 4719 4720void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4721{ 4722 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4723 if (mSyncEvents[i]->type() == type) { 4724 mSyncEvents[i]->trigger(); 4725 mSyncEvents.removeAt(i); 4726 i--; 4727 } 4728 } 4729} 4730 4731// implement VolumeBufferProvider interface 4732 4733uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4734{ 4735 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4736 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4737 uint32_t vlr = mCblk->getVolumeLR(); 4738 uint32_t vl = vlr & 0xFFFF; 4739 uint32_t vr = vlr >> 16; 4740 // track volumes come from shared memory, so can't be trusted and must be clamped 4741 if (vl > MAX_GAIN_INT) { 4742 vl = MAX_GAIN_INT; 4743 } 4744 if (vr > MAX_GAIN_INT) { 4745 vr = MAX_GAIN_INT; 4746 } 4747 // now apply the cached master volume and stream type volume; 4748 // this is trusted but lacks any synchronization or barrier so may be stale 4749 float v = mCachedVolume; 4750 vl *= v; 4751 vr *= v; 4752 // re-combine into U4.16 4753 vlr = (vr << 16) | (vl & 0xFFFF); 4754 // FIXME look at mute, pause, and stop flags 4755 return vlr; 4756} 4757 4758status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4759{ 4760 if (mState == TERMINATED || mState == PAUSED || 4761 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4762 (mState == STOPPED)))) { 4763 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4764 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4765 event->cancel(); 4766 return INVALID_OPERATION; 4767 } 4768 TrackBase::setSyncEvent(event); 4769 return NO_ERROR; 4770} 4771 4772// timed audio tracks 4773 4774sp<AudioFlinger::PlaybackThread::TimedTrack> 4775AudioFlinger::PlaybackThread::TimedTrack::create( 4776 PlaybackThread *thread, 4777 const sp<Client>& client, 4778 audio_stream_type_t streamType, 4779 uint32_t sampleRate, 4780 audio_format_t format, 4781 uint32_t channelMask, 4782 int frameCount, 4783 const sp<IMemory>& sharedBuffer, 4784 int sessionId) { 4785 if (!client->reserveTimedTrack()) 4786 return NULL; 4787 4788 return new TimedTrack( 4789 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4790 sharedBuffer, sessionId); 4791} 4792 4793AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4794 PlaybackThread *thread, 4795 const sp<Client>& client, 4796 audio_stream_type_t streamType, 4797 uint32_t sampleRate, 4798 audio_format_t format, 4799 uint32_t channelMask, 4800 int frameCount, 4801 const sp<IMemory>& sharedBuffer, 4802 int sessionId) 4803 : Track(thread, client, streamType, sampleRate, format, channelMask, 4804 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4805 mQueueHeadInFlight(false), 4806 mTrimQueueHeadOnRelease(false), 4807 mFramesPendingInQueue(0), 4808 mTimedSilenceBuffer(NULL), 4809 mTimedSilenceBufferSize(0), 4810 mTimedAudioOutputOnTime(false), 4811 mMediaTimeTransformValid(false) 4812{ 4813 LocalClock lc; 4814 mLocalTimeFreq = lc.getLocalFreq(); 4815 4816 mLocalTimeToSampleTransform.a_zero = 0; 4817 mLocalTimeToSampleTransform.b_zero = 0; 4818 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4819 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4820 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4821 &mLocalTimeToSampleTransform.a_to_b_denom); 4822 4823 mMediaTimeToSampleTransform.a_zero = 0; 4824 mMediaTimeToSampleTransform.b_zero = 0; 4825 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4826 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4827 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4828 &mMediaTimeToSampleTransform.a_to_b_denom); 4829} 4830 4831AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4832 mClient->releaseTimedTrack(); 4833 delete [] mTimedSilenceBuffer; 4834} 4835 4836status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4837 size_t size, sp<IMemory>* buffer) { 4838 4839 Mutex::Autolock _l(mTimedBufferQueueLock); 4840 4841 trimTimedBufferQueue_l(); 4842 4843 // lazily initialize the shared memory heap for timed buffers 4844 if (mTimedMemoryDealer == NULL) { 4845 const int kTimedBufferHeapSize = 512 << 10; 4846 4847 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4848 "AudioFlingerTimed"); 4849 if (mTimedMemoryDealer == NULL) 4850 return NO_MEMORY; 4851 } 4852 4853 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4854 if (newBuffer == NULL) { 4855 newBuffer = mTimedMemoryDealer->allocate(size); 4856 if (newBuffer == NULL) 4857 return NO_MEMORY; 4858 } 4859 4860 *buffer = newBuffer; 4861 return NO_ERROR; 4862} 4863 4864// caller must hold mTimedBufferQueueLock 4865void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4866 int64_t mediaTimeNow; 4867 { 4868 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4869 if (!mMediaTimeTransformValid) 4870 return; 4871 4872 int64_t targetTimeNow; 4873 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4874 ? mCCHelper.getCommonTime(&targetTimeNow) 4875 : mCCHelper.getLocalTime(&targetTimeNow); 4876 4877 if (OK != res) 4878 return; 4879 4880 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4881 &mediaTimeNow)) { 4882 return; 4883 } 4884 } 4885 4886 size_t trimEnd; 4887 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4888 int64_t bufEnd; 4889 4890 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4891 // We have a next buffer. Just use its PTS as the PTS of the frame 4892 // following the last frame in this buffer. If the stream is sparse 4893 // (ie, there are deliberate gaps left in the stream which should be 4894 // filled with silence by the TimedAudioTrack), then this can result 4895 // in one extra buffer being left un-trimmed when it could have 4896 // been. In general, this is not typical, and we would rather 4897 // optimized away the TS calculation below for the more common case 4898 // where PTSes are contiguous. 4899 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4900 } else { 4901 // We have no next buffer. Compute the PTS of the frame following 4902 // the last frame in this buffer by computing the duration of of 4903 // this frame in media time units and adding it to the PTS of the 4904 // buffer. 4905 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4906 / mCblk->frameSize; 4907 4908 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4909 &bufEnd)) { 4910 ALOGE("Failed to convert frame count of %lld to media time" 4911 " duration" " (scale factor %d/%u) in %s", 4912 frameCount, 4913 mMediaTimeToSampleTransform.a_to_b_numer, 4914 mMediaTimeToSampleTransform.a_to_b_denom, 4915 __PRETTY_FUNCTION__); 4916 break; 4917 } 4918 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4919 } 4920 4921 if (bufEnd > mediaTimeNow) 4922 break; 4923 4924 // Is the buffer we want to use in the middle of a mix operation right 4925 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4926 // from the mixer which should be coming back shortly. 4927 if (!trimEnd && mQueueHeadInFlight) { 4928 mTrimQueueHeadOnRelease = true; 4929 } 4930 } 4931 4932 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4933 if (trimStart < trimEnd) { 4934 // Update the bookkeeping for framesReady() 4935 for (size_t i = trimStart; i < trimEnd; ++i) { 4936 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4937 } 4938 4939 // Now actually remove the buffers from the queue. 4940 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4941 } 4942} 4943 4944void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4945 const char* logTag) { 4946 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4947 "%s called (reason \"%s\"), but timed buffer queue has no" 4948 " elements to trim.", __FUNCTION__, logTag); 4949 4950 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4951 mTimedBufferQueue.removeAt(0); 4952} 4953 4954void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4955 const TimedBuffer& buf, 4956 const char* logTag) { 4957 uint32_t bufBytes = buf.buffer()->size(); 4958 uint32_t consumedAlready = buf.position(); 4959 4960 ALOG_ASSERT(consumedAlready <= bufBytes, 4961 "Bad bookkeeping while updating frames pending. Timed buffer is" 4962 " only %u bytes long, but claims to have consumed %u" 4963 " bytes. (update reason: \"%s\")", 4964 bufBytes, consumedAlready, logTag); 4965 4966 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4967 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4968 "Bad bookkeeping while updating frames pending. Should have at" 4969 " least %u queued frames, but we think we have only %u. (update" 4970 " reason: \"%s\")", 4971 bufFrames, mFramesPendingInQueue, logTag); 4972 4973 mFramesPendingInQueue -= bufFrames; 4974} 4975 4976status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4977 const sp<IMemory>& buffer, int64_t pts) { 4978 4979 { 4980 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4981 if (!mMediaTimeTransformValid) 4982 return INVALID_OPERATION; 4983 } 4984 4985 Mutex::Autolock _l(mTimedBufferQueueLock); 4986 4987 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4988 mFramesPendingInQueue += bufFrames; 4989 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4990 4991 return NO_ERROR; 4992} 4993 4994status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4995 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4996 4997 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4998 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4999 target); 5000 5001 if (!(target == TimedAudioTrack::LOCAL_TIME || 5002 target == TimedAudioTrack::COMMON_TIME)) { 5003 return BAD_VALUE; 5004 } 5005 5006 Mutex::Autolock lock(mMediaTimeTransformLock); 5007 mMediaTimeTransform = xform; 5008 mMediaTimeTransformTarget = target; 5009 mMediaTimeTransformValid = true; 5010 5011 return NO_ERROR; 5012} 5013 5014#define min(a, b) ((a) < (b) ? (a) : (b)) 5015 5016// implementation of getNextBuffer for tracks whose buffers have timestamps 5017status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5018 AudioBufferProvider::Buffer* buffer, int64_t pts) 5019{ 5020 if (pts == AudioBufferProvider::kInvalidPTS) { 5021 buffer->raw = 0; 5022 buffer->frameCount = 0; 5023 mTimedAudioOutputOnTime = false; 5024 return INVALID_OPERATION; 5025 } 5026 5027 Mutex::Autolock _l(mTimedBufferQueueLock); 5028 5029 ALOG_ASSERT(!mQueueHeadInFlight, 5030 "getNextBuffer called without releaseBuffer!"); 5031 5032 while (true) { 5033 5034 // if we have no timed buffers, then fail 5035 if (mTimedBufferQueue.isEmpty()) { 5036 buffer->raw = 0; 5037 buffer->frameCount = 0; 5038 return NOT_ENOUGH_DATA; 5039 } 5040 5041 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5042 5043 // calculate the PTS of the head of the timed buffer queue expressed in 5044 // local time 5045 int64_t headLocalPTS; 5046 { 5047 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5048 5049 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5050 5051 if (mMediaTimeTransform.a_to_b_denom == 0) { 5052 // the transform represents a pause, so yield silence 5053 timedYieldSilence_l(buffer->frameCount, buffer); 5054 return NO_ERROR; 5055 } 5056 5057 int64_t transformedPTS; 5058 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5059 &transformedPTS)) { 5060 // the transform failed. this shouldn't happen, but if it does 5061 // then just drop this buffer 5062 ALOGW("timedGetNextBuffer transform failed"); 5063 buffer->raw = 0; 5064 buffer->frameCount = 0; 5065 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5066 return NO_ERROR; 5067 } 5068 5069 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5070 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5071 &headLocalPTS)) { 5072 buffer->raw = 0; 5073 buffer->frameCount = 0; 5074 return INVALID_OPERATION; 5075 } 5076 } else { 5077 headLocalPTS = transformedPTS; 5078 } 5079 } 5080 5081 // adjust the head buffer's PTS to reflect the portion of the head buffer 5082 // that has already been consumed 5083 int64_t effectivePTS = headLocalPTS + 5084 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5085 5086 // Calculate the delta in samples between the head of the input buffer 5087 // queue and the start of the next output buffer that will be written. 5088 // If the transformation fails because of over or underflow, it means 5089 // that the sample's position in the output stream is so far out of 5090 // whack that it should just be dropped. 5091 int64_t sampleDelta; 5092 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5093 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5094 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5095 " mix"); 5096 continue; 5097 } 5098 if (!mLocalTimeToSampleTransform.doForwardTransform( 5099 (effectivePTS - pts) << 32, &sampleDelta)) { 5100 ALOGV("*** too late during sample rate transform: dropped buffer"); 5101 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5102 continue; 5103 } 5104 5105 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5106 " sampleDelta=[%d.%08x]", 5107 head.pts(), head.position(), pts, 5108 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5109 + (sampleDelta >> 32)), 5110 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5111 5112 // if the delta between the ideal placement for the next input sample and 5113 // the current output position is within this threshold, then we will 5114 // concatenate the next input samples to the previous output 5115 const int64_t kSampleContinuityThreshold = 5116 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5117 5118 // if this is the first buffer of audio that we're emitting from this track 5119 // then it should be almost exactly on time. 5120 const int64_t kSampleStartupThreshold = 1LL << 32; 5121 5122 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5123 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5124 // the next input is close enough to being on time, so concatenate it 5125 // with the last output 5126 timedYieldSamples_l(buffer); 5127 5128 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5129 head.position(), buffer->frameCount); 5130 return NO_ERROR; 5131 } 5132 5133 // Looks like our output is not on time. Reset our on timed status. 5134 // Next time we mix samples from our input queue, then should be within 5135 // the StartupThreshold. 5136 mTimedAudioOutputOnTime = false; 5137 if (sampleDelta > 0) { 5138 // the gap between the current output position and the proper start of 5139 // the next input sample is too big, so fill it with silence 5140 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5141 5142 timedYieldSilence_l(framesUntilNextInput, buffer); 5143 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5144 return NO_ERROR; 5145 } else { 5146 // the next input sample is late 5147 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5148 size_t onTimeSamplePosition = 5149 head.position() + lateFrames * mCblk->frameSize; 5150 5151 if (onTimeSamplePosition > head.buffer()->size()) { 5152 // all the remaining samples in the head are too late, so 5153 // drop it and move on 5154 ALOGV("*** too late: dropped buffer"); 5155 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5156 continue; 5157 } else { 5158 // skip over the late samples 5159 head.setPosition(onTimeSamplePosition); 5160 5161 // yield the available samples 5162 timedYieldSamples_l(buffer); 5163 5164 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5165 return NO_ERROR; 5166 } 5167 } 5168 } 5169} 5170 5171// Yield samples from the timed buffer queue head up to the given output 5172// buffer's capacity. 5173// 5174// Caller must hold mTimedBufferQueueLock 5175void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5176 AudioBufferProvider::Buffer* buffer) { 5177 5178 const TimedBuffer& head = mTimedBufferQueue[0]; 5179 5180 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5181 head.position()); 5182 5183 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5184 mCblk->frameSize); 5185 size_t framesRequested = buffer->frameCount; 5186 buffer->frameCount = min(framesLeftInHead, framesRequested); 5187 5188 mQueueHeadInFlight = true; 5189 mTimedAudioOutputOnTime = true; 5190} 5191 5192// Yield samples of silence up to the given output buffer's capacity 5193// 5194// Caller must hold mTimedBufferQueueLock 5195void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5196 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5197 5198 // lazily allocate a buffer filled with silence 5199 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5200 delete [] mTimedSilenceBuffer; 5201 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5202 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5203 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5204 } 5205 5206 buffer->raw = mTimedSilenceBuffer; 5207 size_t framesRequested = buffer->frameCount; 5208 buffer->frameCount = min(numFrames, framesRequested); 5209 5210 mTimedAudioOutputOnTime = false; 5211} 5212 5213// AudioBufferProvider interface 5214void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5215 AudioBufferProvider::Buffer* buffer) { 5216 5217 Mutex::Autolock _l(mTimedBufferQueueLock); 5218 5219 // If the buffer which was just released is part of the buffer at the head 5220 // of the queue, be sure to update the amt of the buffer which has been 5221 // consumed. If the buffer being returned is not part of the head of the 5222 // queue, its either because the buffer is part of the silence buffer, or 5223 // because the head of the timed queue was trimmed after the mixer called 5224 // getNextBuffer but before the mixer called releaseBuffer. 5225 if (buffer->raw == mTimedSilenceBuffer) { 5226 ALOG_ASSERT(!mQueueHeadInFlight, 5227 "Queue head in flight during release of silence buffer!"); 5228 goto done; 5229 } 5230 5231 ALOG_ASSERT(mQueueHeadInFlight, 5232 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5233 " head in flight."); 5234 5235 if (mTimedBufferQueue.size()) { 5236 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5237 5238 void* start = head.buffer()->pointer(); 5239 void* end = reinterpret_cast<void*>( 5240 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5241 + head.buffer()->size()); 5242 5243 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5244 "released buffer not within the head of the timed buffer" 5245 " queue; qHead = [%p, %p], released buffer = %p", 5246 start, end, buffer->raw); 5247 5248 head.setPosition(head.position() + 5249 (buffer->frameCount * mCblk->frameSize)); 5250 mQueueHeadInFlight = false; 5251 5252 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5253 "Bad bookkeeping during releaseBuffer! Should have at" 5254 " least %u queued frames, but we think we have only %u", 5255 buffer->frameCount, mFramesPendingInQueue); 5256 5257 mFramesPendingInQueue -= buffer->frameCount; 5258 5259 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5260 || mTrimQueueHeadOnRelease) { 5261 trimTimedBufferQueueHead_l("releaseBuffer"); 5262 mTrimQueueHeadOnRelease = false; 5263 } 5264 } else { 5265 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5266 " buffers in the timed buffer queue"); 5267 } 5268 5269done: 5270 buffer->raw = 0; 5271 buffer->frameCount = 0; 5272} 5273 5274size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5275 Mutex::Autolock _l(mTimedBufferQueueLock); 5276 return mFramesPendingInQueue; 5277} 5278 5279AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5280 : mPTS(0), mPosition(0) {} 5281 5282AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5283 const sp<IMemory>& buffer, int64_t pts) 5284 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5285 5286// ---------------------------------------------------------------------------- 5287 5288// RecordTrack constructor must be called with AudioFlinger::mLock held 5289AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5290 RecordThread *thread, 5291 const sp<Client>& client, 5292 uint32_t sampleRate, 5293 audio_format_t format, 5294 uint32_t channelMask, 5295 int frameCount, 5296 int sessionId) 5297 : TrackBase(thread, client, sampleRate, format, 5298 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5299 mOverflow(false) 5300{ 5301 if (mCblk != NULL) { 5302 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5303 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5304 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5305 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5306 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5307 } else { 5308 mCblk->frameSize = sizeof(int8_t); 5309 } 5310 } 5311} 5312 5313AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5314{ 5315 sp<ThreadBase> thread = mThread.promote(); 5316 if (thread != 0) { 5317 AudioSystem::releaseInput(thread->id()); 5318 } 5319} 5320 5321// AudioBufferProvider interface 5322status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5323{ 5324 audio_track_cblk_t* cblk = this->cblk(); 5325 uint32_t framesAvail; 5326 uint32_t framesReq = buffer->frameCount; 5327 5328 // Check if last stepServer failed, try to step now 5329 if (mStepServerFailed) { 5330 if (!step()) goto getNextBuffer_exit; 5331 ALOGV("stepServer recovered"); 5332 mStepServerFailed = false; 5333 } 5334 5335 framesAvail = cblk->framesAvailable_l(); 5336 5337 if (CC_LIKELY(framesAvail)) { 5338 uint32_t s = cblk->server; 5339 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5340 5341 if (framesReq > framesAvail) { 5342 framesReq = framesAvail; 5343 } 5344 if (framesReq > bufferEnd - s) { 5345 framesReq = bufferEnd - s; 5346 } 5347 5348 buffer->raw = getBuffer(s, framesReq); 5349 if (buffer->raw == NULL) goto getNextBuffer_exit; 5350 5351 buffer->frameCount = framesReq; 5352 return NO_ERROR; 5353 } 5354 5355getNextBuffer_exit: 5356 buffer->raw = NULL; 5357 buffer->frameCount = 0; 5358 return NOT_ENOUGH_DATA; 5359} 5360 5361status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5362 int triggerSession) 5363{ 5364 sp<ThreadBase> thread = mThread.promote(); 5365 if (thread != 0) { 5366 RecordThread *recordThread = (RecordThread *)thread.get(); 5367 return recordThread->start(this, event, triggerSession); 5368 } else { 5369 return BAD_VALUE; 5370 } 5371} 5372 5373void AudioFlinger::RecordThread::RecordTrack::stop() 5374{ 5375 sp<ThreadBase> thread = mThread.promote(); 5376 if (thread != 0) { 5377 RecordThread *recordThread = (RecordThread *)thread.get(); 5378 recordThread->stop(this); 5379 TrackBase::reset(); 5380 // Force overrun condition to avoid false overrun callback until first data is 5381 // read from buffer 5382 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5383 } 5384} 5385 5386void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5387{ 5388 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5389 (mClient == 0) ? getpid_cached : mClient->pid(), 5390 mFormat, 5391 mChannelMask, 5392 mSessionId, 5393 mFrameCount, 5394 mState, 5395 mCblk->sampleRate, 5396 mCblk->server, 5397 mCblk->user); 5398} 5399 5400 5401// ---------------------------------------------------------------------------- 5402 5403AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5404 PlaybackThread *playbackThread, 5405 DuplicatingThread *sourceThread, 5406 uint32_t sampleRate, 5407 audio_format_t format, 5408 uint32_t channelMask, 5409 int frameCount) 5410 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5411 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5412 mActive(false), mSourceThread(sourceThread) 5413{ 5414 5415 if (mCblk != NULL) { 5416 mCblk->flags |= CBLK_DIRECTION_OUT; 5417 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5418 mOutBuffer.frameCount = 0; 5419 playbackThread->mTracks.add(this); 5420 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5421 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5422 mCblk, mBuffer, mCblk->buffers, 5423 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5424 } else { 5425 ALOGW("Error creating output track on thread %p", playbackThread); 5426 } 5427} 5428 5429AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5430{ 5431 clearBufferQueue(); 5432} 5433 5434status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5435 int triggerSession) 5436{ 5437 status_t status = Track::start(event, triggerSession); 5438 if (status != NO_ERROR) { 5439 return status; 5440 } 5441 5442 mActive = true; 5443 mRetryCount = 127; 5444 return status; 5445} 5446 5447void AudioFlinger::PlaybackThread::OutputTrack::stop() 5448{ 5449 Track::stop(); 5450 clearBufferQueue(); 5451 mOutBuffer.frameCount = 0; 5452 mActive = false; 5453} 5454 5455bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5456{ 5457 Buffer *pInBuffer; 5458 Buffer inBuffer; 5459 uint32_t channelCount = mChannelCount; 5460 bool outputBufferFull = false; 5461 inBuffer.frameCount = frames; 5462 inBuffer.i16 = data; 5463 5464 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5465 5466 if (!mActive && frames != 0) { 5467 start(); 5468 sp<ThreadBase> thread = mThread.promote(); 5469 if (thread != 0) { 5470 MixerThread *mixerThread = (MixerThread *)thread.get(); 5471 if (mCblk->frameCount > frames){ 5472 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5473 uint32_t startFrames = (mCblk->frameCount - frames); 5474 pInBuffer = new Buffer; 5475 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5476 pInBuffer->frameCount = startFrames; 5477 pInBuffer->i16 = pInBuffer->mBuffer; 5478 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5479 mBufferQueue.add(pInBuffer); 5480 } else { 5481 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5482 } 5483 } 5484 } 5485 } 5486 5487 while (waitTimeLeftMs) { 5488 // First write pending buffers, then new data 5489 if (mBufferQueue.size()) { 5490 pInBuffer = mBufferQueue.itemAt(0); 5491 } else { 5492 pInBuffer = &inBuffer; 5493 } 5494 5495 if (pInBuffer->frameCount == 0) { 5496 break; 5497 } 5498 5499 if (mOutBuffer.frameCount == 0) { 5500 mOutBuffer.frameCount = pInBuffer->frameCount; 5501 nsecs_t startTime = systemTime(); 5502 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5503 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5504 outputBufferFull = true; 5505 break; 5506 } 5507 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5508 if (waitTimeLeftMs >= waitTimeMs) { 5509 waitTimeLeftMs -= waitTimeMs; 5510 } else { 5511 waitTimeLeftMs = 0; 5512 } 5513 } 5514 5515 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5516 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5517 mCblk->stepUser(outFrames); 5518 pInBuffer->frameCount -= outFrames; 5519 pInBuffer->i16 += outFrames * channelCount; 5520 mOutBuffer.frameCount -= outFrames; 5521 mOutBuffer.i16 += outFrames * channelCount; 5522 5523 if (pInBuffer->frameCount == 0) { 5524 if (mBufferQueue.size()) { 5525 mBufferQueue.removeAt(0); 5526 delete [] pInBuffer->mBuffer; 5527 delete pInBuffer; 5528 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5529 } else { 5530 break; 5531 } 5532 } 5533 } 5534 5535 // If we could not write all frames, allocate a buffer and queue it for next time. 5536 if (inBuffer.frameCount) { 5537 sp<ThreadBase> thread = mThread.promote(); 5538 if (thread != 0 && !thread->standby()) { 5539 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5540 pInBuffer = new Buffer; 5541 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5542 pInBuffer->frameCount = inBuffer.frameCount; 5543 pInBuffer->i16 = pInBuffer->mBuffer; 5544 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5545 mBufferQueue.add(pInBuffer); 5546 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5547 } else { 5548 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5549 } 5550 } 5551 } 5552 5553 // Calling write() with a 0 length buffer, means that no more data will be written: 5554 // If no more buffers are pending, fill output track buffer to make sure it is started 5555 // by output mixer. 5556 if (frames == 0 && mBufferQueue.size() == 0) { 5557 if (mCblk->user < mCblk->frameCount) { 5558 frames = mCblk->frameCount - mCblk->user; 5559 pInBuffer = new Buffer; 5560 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5561 pInBuffer->frameCount = frames; 5562 pInBuffer->i16 = pInBuffer->mBuffer; 5563 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5564 mBufferQueue.add(pInBuffer); 5565 } else if (mActive) { 5566 stop(); 5567 } 5568 } 5569 5570 return outputBufferFull; 5571} 5572 5573status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5574{ 5575 int active; 5576 status_t result; 5577 audio_track_cblk_t* cblk = mCblk; 5578 uint32_t framesReq = buffer->frameCount; 5579 5580// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5581 buffer->frameCount = 0; 5582 5583 uint32_t framesAvail = cblk->framesAvailable(); 5584 5585 5586 if (framesAvail == 0) { 5587 Mutex::Autolock _l(cblk->lock); 5588 goto start_loop_here; 5589 while (framesAvail == 0) { 5590 active = mActive; 5591 if (CC_UNLIKELY(!active)) { 5592 ALOGV("Not active and NO_MORE_BUFFERS"); 5593 return NO_MORE_BUFFERS; 5594 } 5595 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5596 if (result != NO_ERROR) { 5597 return NO_MORE_BUFFERS; 5598 } 5599 // read the server count again 5600 start_loop_here: 5601 framesAvail = cblk->framesAvailable_l(); 5602 } 5603 } 5604 5605// if (framesAvail < framesReq) { 5606// return NO_MORE_BUFFERS; 5607// } 5608 5609 if (framesReq > framesAvail) { 5610 framesReq = framesAvail; 5611 } 5612 5613 uint32_t u = cblk->user; 5614 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5615 5616 if (framesReq > bufferEnd - u) { 5617 framesReq = bufferEnd - u; 5618 } 5619 5620 buffer->frameCount = framesReq; 5621 buffer->raw = (void *)cblk->buffer(u); 5622 return NO_ERROR; 5623} 5624 5625 5626void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5627{ 5628 size_t size = mBufferQueue.size(); 5629 5630 for (size_t i = 0; i < size; i++) { 5631 Buffer *pBuffer = mBufferQueue.itemAt(i); 5632 delete [] pBuffer->mBuffer; 5633 delete pBuffer; 5634 } 5635 mBufferQueue.clear(); 5636} 5637 5638// ---------------------------------------------------------------------------- 5639 5640AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5641 : RefBase(), 5642 mAudioFlinger(audioFlinger), 5643 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5644 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5645 mPid(pid), 5646 mTimedTrackCount(0) 5647{ 5648 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5649} 5650 5651// Client destructor must be called with AudioFlinger::mLock held 5652AudioFlinger::Client::~Client() 5653{ 5654 mAudioFlinger->removeClient_l(mPid); 5655} 5656 5657sp<MemoryDealer> AudioFlinger::Client::heap() const 5658{ 5659 return mMemoryDealer; 5660} 5661 5662// Reserve one of the limited slots for a timed audio track associated 5663// with this client 5664bool AudioFlinger::Client::reserveTimedTrack() 5665{ 5666 const int kMaxTimedTracksPerClient = 4; 5667 5668 Mutex::Autolock _l(mTimedTrackLock); 5669 5670 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5671 ALOGW("can not create timed track - pid %d has exceeded the limit", 5672 mPid); 5673 return false; 5674 } 5675 5676 mTimedTrackCount++; 5677 return true; 5678} 5679 5680// Release a slot for a timed audio track 5681void AudioFlinger::Client::releaseTimedTrack() 5682{ 5683 Mutex::Autolock _l(mTimedTrackLock); 5684 mTimedTrackCount--; 5685} 5686 5687// ---------------------------------------------------------------------------- 5688 5689AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5690 const sp<IAudioFlingerClient>& client, 5691 pid_t pid) 5692 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5693{ 5694} 5695 5696AudioFlinger::NotificationClient::~NotificationClient() 5697{ 5698} 5699 5700void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5701{ 5702 sp<NotificationClient> keep(this); 5703 mAudioFlinger->removeNotificationClient(mPid); 5704} 5705 5706// ---------------------------------------------------------------------------- 5707 5708AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5709 : BnAudioTrack(), 5710 mTrack(track) 5711{ 5712} 5713 5714AudioFlinger::TrackHandle::~TrackHandle() { 5715 // just stop the track on deletion, associated resources 5716 // will be freed from the main thread once all pending buffers have 5717 // been played. Unless it's not in the active track list, in which 5718 // case we free everything now... 5719 mTrack->destroy(); 5720} 5721 5722sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5723 return mTrack->getCblk(); 5724} 5725 5726status_t AudioFlinger::TrackHandle::start() { 5727 return mTrack->start(); 5728} 5729 5730void AudioFlinger::TrackHandle::stop() { 5731 mTrack->stop(); 5732} 5733 5734void AudioFlinger::TrackHandle::flush() { 5735 mTrack->flush(); 5736} 5737 5738void AudioFlinger::TrackHandle::mute(bool e) { 5739 mTrack->mute(e); 5740} 5741 5742void AudioFlinger::TrackHandle::pause() { 5743 mTrack->pause(); 5744} 5745 5746status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5747{ 5748 return mTrack->attachAuxEffect(EffectId); 5749} 5750 5751status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5752 sp<IMemory>* buffer) { 5753 if (!mTrack->isTimedTrack()) 5754 return INVALID_OPERATION; 5755 5756 PlaybackThread::TimedTrack* tt = 5757 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5758 return tt->allocateTimedBuffer(size, buffer); 5759} 5760 5761status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5762 int64_t pts) { 5763 if (!mTrack->isTimedTrack()) 5764 return INVALID_OPERATION; 5765 5766 PlaybackThread::TimedTrack* tt = 5767 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5768 return tt->queueTimedBuffer(buffer, pts); 5769} 5770 5771status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5772 const LinearTransform& xform, int target) { 5773 5774 if (!mTrack->isTimedTrack()) 5775 return INVALID_OPERATION; 5776 5777 PlaybackThread::TimedTrack* tt = 5778 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5779 return tt->setMediaTimeTransform( 5780 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5781} 5782 5783status_t AudioFlinger::TrackHandle::onTransact( 5784 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5785{ 5786 return BnAudioTrack::onTransact(code, data, reply, flags); 5787} 5788 5789// ---------------------------------------------------------------------------- 5790 5791sp<IAudioRecord> AudioFlinger::openRecord( 5792 pid_t pid, 5793 audio_io_handle_t input, 5794 uint32_t sampleRate, 5795 audio_format_t format, 5796 uint32_t channelMask, 5797 int frameCount, 5798 IAudioFlinger::track_flags_t flags, 5799 int *sessionId, 5800 status_t *status) 5801{ 5802 sp<RecordThread::RecordTrack> recordTrack; 5803 sp<RecordHandle> recordHandle; 5804 sp<Client> client; 5805 status_t lStatus; 5806 RecordThread *thread; 5807 size_t inFrameCount; 5808 int lSessionId; 5809 5810 // check calling permissions 5811 if (!recordingAllowed()) { 5812 lStatus = PERMISSION_DENIED; 5813 goto Exit; 5814 } 5815 5816 // add client to list 5817 { // scope for mLock 5818 Mutex::Autolock _l(mLock); 5819 thread = checkRecordThread_l(input); 5820 if (thread == NULL) { 5821 lStatus = BAD_VALUE; 5822 goto Exit; 5823 } 5824 5825 client = registerPid_l(pid); 5826 5827 // If no audio session id is provided, create one here 5828 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5829 lSessionId = *sessionId; 5830 } else { 5831 lSessionId = nextUniqueId(); 5832 if (sessionId != NULL) { 5833 *sessionId = lSessionId; 5834 } 5835 } 5836 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5837 recordTrack = thread->createRecordTrack_l(client, 5838 sampleRate, 5839 format, 5840 channelMask, 5841 frameCount, 5842 lSessionId, 5843 &lStatus); 5844 } 5845 if (lStatus != NO_ERROR) { 5846 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5847 // destructor is called by the TrackBase destructor with mLock held 5848 client.clear(); 5849 recordTrack.clear(); 5850 goto Exit; 5851 } 5852 5853 // return to handle to client 5854 recordHandle = new RecordHandle(recordTrack); 5855 lStatus = NO_ERROR; 5856 5857Exit: 5858 if (status) { 5859 *status = lStatus; 5860 } 5861 return recordHandle; 5862} 5863 5864// ---------------------------------------------------------------------------- 5865 5866AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5867 : BnAudioRecord(), 5868 mRecordTrack(recordTrack) 5869{ 5870} 5871 5872AudioFlinger::RecordHandle::~RecordHandle() { 5873 stop(); 5874} 5875 5876sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5877 return mRecordTrack->getCblk(); 5878} 5879 5880status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5881 ALOGV("RecordHandle::start()"); 5882 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5883} 5884 5885void AudioFlinger::RecordHandle::stop() { 5886 ALOGV("RecordHandle::stop()"); 5887 mRecordTrack->stop(); 5888} 5889 5890status_t AudioFlinger::RecordHandle::onTransact( 5891 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5892{ 5893 return BnAudioRecord::onTransact(code, data, reply, flags); 5894} 5895 5896// ---------------------------------------------------------------------------- 5897 5898AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5899 AudioStreamIn *input, 5900 uint32_t sampleRate, 5901 uint32_t channels, 5902 audio_io_handle_t id, 5903 uint32_t device) : 5904 ThreadBase(audioFlinger, id, device, RECORD), 5905 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5906 // mRsmpInIndex and mInputBytes set by readInputParameters() 5907 mReqChannelCount(popcount(channels)), 5908 mReqSampleRate(sampleRate) 5909 // mBytesRead is only meaningful while active, and so is cleared in start() 5910 // (but might be better to also clear here for dump?) 5911{ 5912 snprintf(mName, kNameLength, "AudioIn_%X", id); 5913 5914 readInputParameters(); 5915} 5916 5917 5918AudioFlinger::RecordThread::~RecordThread() 5919{ 5920 delete[] mRsmpInBuffer; 5921 delete mResampler; 5922 delete[] mRsmpOutBuffer; 5923} 5924 5925void AudioFlinger::RecordThread::onFirstRef() 5926{ 5927 run(mName, PRIORITY_URGENT_AUDIO); 5928} 5929 5930status_t AudioFlinger::RecordThread::readyToRun() 5931{ 5932 status_t status = initCheck(); 5933 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5934 return status; 5935} 5936 5937bool AudioFlinger::RecordThread::threadLoop() 5938{ 5939 AudioBufferProvider::Buffer buffer; 5940 sp<RecordTrack> activeTrack; 5941 Vector< sp<EffectChain> > effectChains; 5942 5943 nsecs_t lastWarning = 0; 5944 5945 acquireWakeLock(); 5946 5947 // start recording 5948 while (!exitPending()) { 5949 5950 processConfigEvents(); 5951 5952 { // scope for mLock 5953 Mutex::Autolock _l(mLock); 5954 checkForNewParameters_l(); 5955 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5956 if (!mStandby) { 5957 mInput->stream->common.standby(&mInput->stream->common); 5958 mStandby = true; 5959 } 5960 5961 if (exitPending()) break; 5962 5963 releaseWakeLock_l(); 5964 ALOGV("RecordThread: loop stopping"); 5965 // go to sleep 5966 mWaitWorkCV.wait(mLock); 5967 ALOGV("RecordThread: loop starting"); 5968 acquireWakeLock_l(); 5969 continue; 5970 } 5971 if (mActiveTrack != 0) { 5972 if (mActiveTrack->mState == TrackBase::PAUSING) { 5973 if (!mStandby) { 5974 mInput->stream->common.standby(&mInput->stream->common); 5975 mStandby = true; 5976 } 5977 mActiveTrack.clear(); 5978 mStartStopCond.broadcast(); 5979 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5980 if (mReqChannelCount != mActiveTrack->channelCount()) { 5981 mActiveTrack.clear(); 5982 mStartStopCond.broadcast(); 5983 } else if (mBytesRead != 0) { 5984 // record start succeeds only if first read from audio input 5985 // succeeds 5986 if (mBytesRead > 0) { 5987 mActiveTrack->mState = TrackBase::ACTIVE; 5988 } else { 5989 mActiveTrack.clear(); 5990 } 5991 mStartStopCond.broadcast(); 5992 } 5993 mStandby = false; 5994 } 5995 } 5996 lockEffectChains_l(effectChains); 5997 } 5998 5999 if (mActiveTrack != 0) { 6000 if (mActiveTrack->mState != TrackBase::ACTIVE && 6001 mActiveTrack->mState != TrackBase::RESUMING) { 6002 unlockEffectChains(effectChains); 6003 usleep(kRecordThreadSleepUs); 6004 continue; 6005 } 6006 for (size_t i = 0; i < effectChains.size(); i ++) { 6007 effectChains[i]->process_l(); 6008 } 6009 6010 buffer.frameCount = mFrameCount; 6011 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6012 size_t framesOut = buffer.frameCount; 6013 if (mResampler == NULL) { 6014 // no resampling 6015 while (framesOut) { 6016 size_t framesIn = mFrameCount - mRsmpInIndex; 6017 if (framesIn) { 6018 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6019 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6020 if (framesIn > framesOut) 6021 framesIn = framesOut; 6022 mRsmpInIndex += framesIn; 6023 framesOut -= framesIn; 6024 if ((int)mChannelCount == mReqChannelCount || 6025 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6026 memcpy(dst, src, framesIn * mFrameSize); 6027 } else { 6028 int16_t *src16 = (int16_t *)src; 6029 int16_t *dst16 = (int16_t *)dst; 6030 if (mChannelCount == 1) { 6031 while (framesIn--) { 6032 *dst16++ = *src16; 6033 *dst16++ = *src16++; 6034 } 6035 } else { 6036 while (framesIn--) { 6037 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6038 src16 += 2; 6039 } 6040 } 6041 } 6042 } 6043 if (framesOut && mFrameCount == mRsmpInIndex) { 6044 if (framesOut == mFrameCount && 6045 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6046 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6047 framesOut = 0; 6048 } else { 6049 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6050 mRsmpInIndex = 0; 6051 } 6052 if (mBytesRead < 0) { 6053 ALOGE("Error reading audio input"); 6054 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6055 // Force input into standby so that it tries to 6056 // recover at next read attempt 6057 mInput->stream->common.standby(&mInput->stream->common); 6058 usleep(kRecordThreadSleepUs); 6059 } 6060 mRsmpInIndex = mFrameCount; 6061 framesOut = 0; 6062 buffer.frameCount = 0; 6063 } 6064 } 6065 } 6066 } else { 6067 // resampling 6068 6069 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6070 // alter output frame count as if we were expecting stereo samples 6071 if (mChannelCount == 1 && mReqChannelCount == 1) { 6072 framesOut >>= 1; 6073 } 6074 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6075 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6076 // are 32 bit aligned which should be always true. 6077 if (mChannelCount == 2 && mReqChannelCount == 1) { 6078 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6079 // the resampler always outputs stereo samples: do post stereo to mono conversion 6080 int16_t *src = (int16_t *)mRsmpOutBuffer; 6081 int16_t *dst = buffer.i16; 6082 while (framesOut--) { 6083 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6084 src += 2; 6085 } 6086 } else { 6087 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6088 } 6089 6090 } 6091 if (mFramestoDrop == 0) { 6092 mActiveTrack->releaseBuffer(&buffer); 6093 } else { 6094 if (mFramestoDrop > 0) { 6095 mFramestoDrop -= buffer.frameCount; 6096 if (mFramestoDrop <= 0) { 6097 clearSyncStartEvent(); 6098 } 6099 } else { 6100 mFramestoDrop += buffer.frameCount; 6101 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6102 mSyncStartEvent->isCancelled()) { 6103 ALOGW("Synced record %s, session %d, trigger session %d", 6104 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6105 mActiveTrack->sessionId(), 6106 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6107 clearSyncStartEvent(); 6108 } 6109 } 6110 } 6111 mActiveTrack->overflow(); 6112 } 6113 // client isn't retrieving buffers fast enough 6114 else { 6115 if (!mActiveTrack->setOverflow()) { 6116 nsecs_t now = systemTime(); 6117 if ((now - lastWarning) > kWarningThrottleNs) { 6118 ALOGW("RecordThread: buffer overflow"); 6119 lastWarning = now; 6120 } 6121 } 6122 // Release the processor for a while before asking for a new buffer. 6123 // This will give the application more chance to read from the buffer and 6124 // clear the overflow. 6125 usleep(kRecordThreadSleepUs); 6126 } 6127 } 6128 // enable changes in effect chain 6129 unlockEffectChains(effectChains); 6130 effectChains.clear(); 6131 } 6132 6133 if (!mStandby) { 6134 mInput->stream->common.standby(&mInput->stream->common); 6135 } 6136 mActiveTrack.clear(); 6137 6138 mStartStopCond.broadcast(); 6139 6140 releaseWakeLock(); 6141 6142 ALOGV("RecordThread %p exiting", this); 6143 return false; 6144} 6145 6146 6147sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6148 const sp<AudioFlinger::Client>& client, 6149 uint32_t sampleRate, 6150 audio_format_t format, 6151 int channelMask, 6152 int frameCount, 6153 int sessionId, 6154 status_t *status) 6155{ 6156 sp<RecordTrack> track; 6157 status_t lStatus; 6158 6159 lStatus = initCheck(); 6160 if (lStatus != NO_ERROR) { 6161 ALOGE("Audio driver not initialized."); 6162 goto Exit; 6163 } 6164 6165 { // scope for mLock 6166 Mutex::Autolock _l(mLock); 6167 6168 track = new RecordTrack(this, client, sampleRate, 6169 format, channelMask, frameCount, sessionId); 6170 6171 if (track->getCblk() == 0) { 6172 lStatus = NO_MEMORY; 6173 goto Exit; 6174 } 6175 6176 mTrack = track.get(); 6177 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6178 bool suspend = audio_is_bluetooth_sco_device( 6179 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6180 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6181 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6182 } 6183 lStatus = NO_ERROR; 6184 6185Exit: 6186 if (status) { 6187 *status = lStatus; 6188 } 6189 return track; 6190} 6191 6192status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6193 AudioSystem::sync_event_t event, 6194 int triggerSession) 6195{ 6196 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6197 sp<ThreadBase> strongMe = this; 6198 status_t status = NO_ERROR; 6199 6200 if (event == AudioSystem::SYNC_EVENT_NONE) { 6201 clearSyncStartEvent(); 6202 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6203 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6204 triggerSession, 6205 recordTrack->sessionId(), 6206 syncStartEventCallback, 6207 this); 6208 // Sync event can be cancelled by the trigger session if the track is not in a 6209 // compatible state in which case we start record immediately 6210 if (mSyncStartEvent->isCancelled()) { 6211 clearSyncStartEvent(); 6212 } else { 6213 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6214 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6215 } 6216 } 6217 6218 { 6219 AutoMutex lock(mLock); 6220 if (mActiveTrack != 0) { 6221 if (recordTrack != mActiveTrack.get()) { 6222 status = -EBUSY; 6223 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6224 mActiveTrack->mState = TrackBase::ACTIVE; 6225 } 6226 return status; 6227 } 6228 6229 recordTrack->mState = TrackBase::IDLE; 6230 mActiveTrack = recordTrack; 6231 mLock.unlock(); 6232 status_t status = AudioSystem::startInput(mId); 6233 mLock.lock(); 6234 if (status != NO_ERROR) { 6235 mActiveTrack.clear(); 6236 clearSyncStartEvent(); 6237 return status; 6238 } 6239 mRsmpInIndex = mFrameCount; 6240 mBytesRead = 0; 6241 if (mResampler != NULL) { 6242 mResampler->reset(); 6243 } 6244 mActiveTrack->mState = TrackBase::RESUMING; 6245 // signal thread to start 6246 ALOGV("Signal record thread"); 6247 mWaitWorkCV.signal(); 6248 // do not wait for mStartStopCond if exiting 6249 if (exitPending()) { 6250 mActiveTrack.clear(); 6251 status = INVALID_OPERATION; 6252 goto startError; 6253 } 6254 mStartStopCond.wait(mLock); 6255 if (mActiveTrack == 0) { 6256 ALOGV("Record failed to start"); 6257 status = BAD_VALUE; 6258 goto startError; 6259 } 6260 ALOGV("Record started OK"); 6261 return status; 6262 } 6263startError: 6264 AudioSystem::stopInput(mId); 6265 clearSyncStartEvent(); 6266 return status; 6267} 6268 6269void AudioFlinger::RecordThread::clearSyncStartEvent() 6270{ 6271 if (mSyncStartEvent != 0) { 6272 mSyncStartEvent->cancel(); 6273 } 6274 mSyncStartEvent.clear(); 6275 mFramestoDrop = 0; 6276} 6277 6278void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6279{ 6280 sp<SyncEvent> strongEvent = event.promote(); 6281 6282 if (strongEvent != 0) { 6283 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6284 me->handleSyncStartEvent(strongEvent); 6285 } 6286} 6287 6288void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6289{ 6290 if (event == mSyncStartEvent) { 6291 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6292 // from audio HAL 6293 mFramestoDrop = mFrameCount * 2; 6294 } 6295} 6296 6297void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6298 ALOGV("RecordThread::stop"); 6299 sp<ThreadBase> strongMe = this; 6300 { 6301 AutoMutex lock(mLock); 6302 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6303 mActiveTrack->mState = TrackBase::PAUSING; 6304 // do not wait for mStartStopCond if exiting 6305 if (exitPending()) { 6306 return; 6307 } 6308 mStartStopCond.wait(mLock); 6309 // if we have been restarted, recordTrack == mActiveTrack.get() here 6310 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6311 mLock.unlock(); 6312 AudioSystem::stopInput(mId); 6313 mLock.lock(); 6314 ALOGV("Record stopped OK"); 6315 } 6316 } 6317 } 6318} 6319 6320bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6321{ 6322 return false; 6323} 6324 6325status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6326{ 6327 if (!isValidSyncEvent(event)) { 6328 return BAD_VALUE; 6329 } 6330 6331 Mutex::Autolock _l(mLock); 6332 6333 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6334 mTrack->setSyncEvent(event); 6335 return NO_ERROR; 6336 } 6337 return NAME_NOT_FOUND; 6338} 6339 6340status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6341{ 6342 const size_t SIZE = 256; 6343 char buffer[SIZE]; 6344 String8 result; 6345 6346 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6347 result.append(buffer); 6348 6349 if (mActiveTrack != 0) { 6350 result.append("Active Track:\n"); 6351 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6352 mActiveTrack->dump(buffer, SIZE); 6353 result.append(buffer); 6354 6355 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6356 result.append(buffer); 6357 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6358 result.append(buffer); 6359 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6360 result.append(buffer); 6361 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6362 result.append(buffer); 6363 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6364 result.append(buffer); 6365 6366 6367 } else { 6368 result.append("No record client\n"); 6369 } 6370 write(fd, result.string(), result.size()); 6371 6372 dumpBase(fd, args); 6373 dumpEffectChains(fd, args); 6374 6375 return NO_ERROR; 6376} 6377 6378// AudioBufferProvider interface 6379status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6380{ 6381 size_t framesReq = buffer->frameCount; 6382 size_t framesReady = mFrameCount - mRsmpInIndex; 6383 int channelCount; 6384 6385 if (framesReady == 0) { 6386 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6387 if (mBytesRead < 0) { 6388 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6389 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6390 // Force input into standby so that it tries to 6391 // recover at next read attempt 6392 mInput->stream->common.standby(&mInput->stream->common); 6393 usleep(kRecordThreadSleepUs); 6394 } 6395 buffer->raw = NULL; 6396 buffer->frameCount = 0; 6397 return NOT_ENOUGH_DATA; 6398 } 6399 mRsmpInIndex = 0; 6400 framesReady = mFrameCount; 6401 } 6402 6403 if (framesReq > framesReady) { 6404 framesReq = framesReady; 6405 } 6406 6407 if (mChannelCount == 1 && mReqChannelCount == 2) { 6408 channelCount = 1; 6409 } else { 6410 channelCount = 2; 6411 } 6412 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6413 buffer->frameCount = framesReq; 6414 return NO_ERROR; 6415} 6416 6417// AudioBufferProvider interface 6418void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6419{ 6420 mRsmpInIndex += buffer->frameCount; 6421 buffer->frameCount = 0; 6422} 6423 6424bool AudioFlinger::RecordThread::checkForNewParameters_l() 6425{ 6426 bool reconfig = false; 6427 6428 while (!mNewParameters.isEmpty()) { 6429 status_t status = NO_ERROR; 6430 String8 keyValuePair = mNewParameters[0]; 6431 AudioParameter param = AudioParameter(keyValuePair); 6432 int value; 6433 audio_format_t reqFormat = mFormat; 6434 int reqSamplingRate = mReqSampleRate; 6435 int reqChannelCount = mReqChannelCount; 6436 6437 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6438 reqSamplingRate = value; 6439 reconfig = true; 6440 } 6441 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6442 reqFormat = (audio_format_t) value; 6443 reconfig = true; 6444 } 6445 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6446 reqChannelCount = popcount(value); 6447 reconfig = true; 6448 } 6449 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6450 // do not accept frame count changes if tracks are open as the track buffer 6451 // size depends on frame count and correct behavior would not be guaranteed 6452 // if frame count is changed after track creation 6453 if (mActiveTrack != 0) { 6454 status = INVALID_OPERATION; 6455 } else { 6456 reconfig = true; 6457 } 6458 } 6459 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6460 // forward device change to effects that have requested to be 6461 // aware of attached audio device. 6462 for (size_t i = 0; i < mEffectChains.size(); i++) { 6463 mEffectChains[i]->setDevice_l(value); 6464 } 6465 // store input device and output device but do not forward output device to audio HAL. 6466 // Note that status is ignored by the caller for output device 6467 // (see AudioFlinger::setParameters() 6468 if (value & AUDIO_DEVICE_OUT_ALL) { 6469 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6470 status = BAD_VALUE; 6471 } else { 6472 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6473 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6474 if (mTrack != NULL) { 6475 bool suspend = audio_is_bluetooth_sco_device( 6476 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6477 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6478 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6479 } 6480 } 6481 mDevice |= (uint32_t)value; 6482 } 6483 if (status == NO_ERROR) { 6484 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6485 if (status == INVALID_OPERATION) { 6486 mInput->stream->common.standby(&mInput->stream->common); 6487 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6488 keyValuePair.string()); 6489 } 6490 if (reconfig) { 6491 if (status == BAD_VALUE && 6492 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6493 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6494 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6495 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6496 (reqChannelCount <= FCC_2)) { 6497 status = NO_ERROR; 6498 } 6499 if (status == NO_ERROR) { 6500 readInputParameters(); 6501 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6502 } 6503 } 6504 } 6505 6506 mNewParameters.removeAt(0); 6507 6508 mParamStatus = status; 6509 mParamCond.signal(); 6510 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6511 // already timed out waiting for the status and will never signal the condition. 6512 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6513 } 6514 return reconfig; 6515} 6516 6517String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6518{ 6519 char *s; 6520 String8 out_s8 = String8(); 6521 6522 Mutex::Autolock _l(mLock); 6523 if (initCheck() != NO_ERROR) { 6524 return out_s8; 6525 } 6526 6527 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6528 out_s8 = String8(s); 6529 free(s); 6530 return out_s8; 6531} 6532 6533void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6534 AudioSystem::OutputDescriptor desc; 6535 void *param2 = NULL; 6536 6537 switch (event) { 6538 case AudioSystem::INPUT_OPENED: 6539 case AudioSystem::INPUT_CONFIG_CHANGED: 6540 desc.channels = mChannelMask; 6541 desc.samplingRate = mSampleRate; 6542 desc.format = mFormat; 6543 desc.frameCount = mFrameCount; 6544 desc.latency = 0; 6545 param2 = &desc; 6546 break; 6547 6548 case AudioSystem::INPUT_CLOSED: 6549 default: 6550 break; 6551 } 6552 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6553} 6554 6555void AudioFlinger::RecordThread::readInputParameters() 6556{ 6557 delete mRsmpInBuffer; 6558 // mRsmpInBuffer is always assigned a new[] below 6559 delete mRsmpOutBuffer; 6560 mRsmpOutBuffer = NULL; 6561 delete mResampler; 6562 mResampler = NULL; 6563 6564 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6565 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6566 mChannelCount = (uint16_t)popcount(mChannelMask); 6567 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6568 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6569 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6570 mFrameCount = mInputBytes / mFrameSize; 6571 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6572 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6573 6574 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6575 { 6576 int channelCount; 6577 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6578 // stereo to mono post process as the resampler always outputs stereo. 6579 if (mChannelCount == 1 && mReqChannelCount == 2) { 6580 channelCount = 1; 6581 } else { 6582 channelCount = 2; 6583 } 6584 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6585 mResampler->setSampleRate(mSampleRate); 6586 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6587 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6588 6589 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6590 if (mChannelCount == 1 && mReqChannelCount == 1) { 6591 mFrameCount >>= 1; 6592 } 6593 6594 } 6595 mRsmpInIndex = mFrameCount; 6596} 6597 6598unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6599{ 6600 Mutex::Autolock _l(mLock); 6601 if (initCheck() != NO_ERROR) { 6602 return 0; 6603 } 6604 6605 return mInput->stream->get_input_frames_lost(mInput->stream); 6606} 6607 6608uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6609{ 6610 Mutex::Autolock _l(mLock); 6611 uint32_t result = 0; 6612 if (getEffectChain_l(sessionId) != 0) { 6613 result = EFFECT_SESSION; 6614 } 6615 6616 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6617 result |= TRACK_SESSION; 6618 } 6619 6620 return result; 6621} 6622 6623AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6624{ 6625 Mutex::Autolock _l(mLock); 6626 return mTrack; 6627} 6628 6629AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6630{ 6631 Mutex::Autolock _l(mLock); 6632 return mInput; 6633} 6634 6635AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6636{ 6637 Mutex::Autolock _l(mLock); 6638 AudioStreamIn *input = mInput; 6639 mInput = NULL; 6640 return input; 6641} 6642 6643// this method must always be called either with ThreadBase mLock held or inside the thread loop 6644audio_stream_t* AudioFlinger::RecordThread::stream() const 6645{ 6646 if (mInput == NULL) { 6647 return NULL; 6648 } 6649 return &mInput->stream->common; 6650} 6651 6652 6653// ---------------------------------------------------------------------------- 6654 6655audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6656{ 6657 if (!settingsAllowed()) { 6658 return 0; 6659 } 6660 Mutex::Autolock _l(mLock); 6661 return loadHwModule_l(name); 6662} 6663 6664// loadHwModule_l() must be called with AudioFlinger::mLock held 6665audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6666{ 6667 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6668 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6669 ALOGW("loadHwModule() module %s already loaded", name); 6670 return mAudioHwDevs.keyAt(i); 6671 } 6672 } 6673 6674 audio_hw_device_t *dev; 6675 6676 int rc = load_audio_interface(name, &dev); 6677 if (rc) { 6678 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6679 return 0; 6680 } 6681 6682 mHardwareStatus = AUDIO_HW_INIT; 6683 rc = dev->init_check(dev); 6684 mHardwareStatus = AUDIO_HW_IDLE; 6685 if (rc) { 6686 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6687 return 0; 6688 } 6689 6690 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6691 (NULL != dev->set_master_volume)) { 6692 AutoMutex lock(mHardwareLock); 6693 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6694 dev->set_master_volume(dev, mMasterVolume); 6695 mHardwareStatus = AUDIO_HW_IDLE; 6696 } 6697 6698 audio_module_handle_t handle = nextUniqueId(); 6699 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6700 6701 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6702 name, dev->common.module->name, dev->common.module->id, handle); 6703 6704 return handle; 6705 6706} 6707 6708audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6709 audio_devices_t *pDevices, 6710 uint32_t *pSamplingRate, 6711 audio_format_t *pFormat, 6712 audio_channel_mask_t *pChannelMask, 6713 uint32_t *pLatencyMs, 6714 audio_output_flags_t flags) 6715{ 6716 status_t status; 6717 PlaybackThread *thread = NULL; 6718 struct audio_config config = { 6719 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6720 channel_mask: pChannelMask ? *pChannelMask : 0, 6721 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6722 }; 6723 audio_stream_out_t *outStream = NULL; 6724 audio_hw_device_t *outHwDev; 6725 6726 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6727 module, 6728 (pDevices != NULL) ? (int)*pDevices : 0, 6729 config.sample_rate, 6730 config.format, 6731 config.channel_mask, 6732 flags); 6733 6734 if (pDevices == NULL || *pDevices == 0) { 6735 return 0; 6736 } 6737 6738 Mutex::Autolock _l(mLock); 6739 6740 outHwDev = findSuitableHwDev_l(module, *pDevices); 6741 if (outHwDev == NULL) 6742 return 0; 6743 6744 audio_io_handle_t id = nextUniqueId(); 6745 6746 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6747 6748 status = outHwDev->open_output_stream(outHwDev, 6749 id, 6750 *pDevices, 6751 (audio_output_flags_t)flags, 6752 &config, 6753 &outStream); 6754 6755 mHardwareStatus = AUDIO_HW_IDLE; 6756 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6757 outStream, 6758 config.sample_rate, 6759 config.format, 6760 config.channel_mask, 6761 status); 6762 6763 if (status == NO_ERROR && outStream != NULL) { 6764 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6765 6766 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6767 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6768 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6769 thread = new DirectOutputThread(this, output, id, *pDevices); 6770 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6771 } else { 6772 thread = new MixerThread(this, output, id, *pDevices); 6773 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6774 } 6775 mPlaybackThreads.add(id, thread); 6776 6777 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6778 if (pFormat != NULL) *pFormat = config.format; 6779 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6780 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6781 6782 // notify client processes of the new output creation 6783 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6784 6785 // the first primary output opened designates the primary hw device 6786 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6787 ALOGI("Using module %d has the primary audio interface", module); 6788 mPrimaryHardwareDev = outHwDev; 6789 6790 AutoMutex lock(mHardwareLock); 6791 mHardwareStatus = AUDIO_HW_SET_MODE; 6792 outHwDev->set_mode(outHwDev, mMode); 6793 6794 // Determine the level of master volume support the primary audio HAL has, 6795 // and set the initial master volume at the same time. 6796 float initialVolume = 1.0; 6797 mMasterVolumeSupportLvl = MVS_NONE; 6798 6799 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6800 if ((NULL != outHwDev->get_master_volume) && 6801 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6802 mMasterVolumeSupportLvl = MVS_FULL; 6803 } else { 6804 mMasterVolumeSupportLvl = MVS_SETONLY; 6805 initialVolume = 1.0; 6806 } 6807 6808 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6809 if ((NULL == outHwDev->set_master_volume) || 6810 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6811 mMasterVolumeSupportLvl = MVS_NONE; 6812 } 6813 // now that we have a primary device, initialize master volume on other devices 6814 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6815 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6816 6817 if ((dev != mPrimaryHardwareDev) && 6818 (NULL != dev->set_master_volume)) { 6819 dev->set_master_volume(dev, initialVolume); 6820 } 6821 } 6822 mHardwareStatus = AUDIO_HW_IDLE; 6823 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6824 ? initialVolume 6825 : 1.0; 6826 mMasterVolume = initialVolume; 6827 } 6828 return id; 6829 } 6830 6831 return 0; 6832} 6833 6834audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6835 audio_io_handle_t output2) 6836{ 6837 Mutex::Autolock _l(mLock); 6838 MixerThread *thread1 = checkMixerThread_l(output1); 6839 MixerThread *thread2 = checkMixerThread_l(output2); 6840 6841 if (thread1 == NULL || thread2 == NULL) { 6842 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6843 return 0; 6844 } 6845 6846 audio_io_handle_t id = nextUniqueId(); 6847 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6848 thread->addOutputTrack(thread2); 6849 mPlaybackThreads.add(id, thread); 6850 // notify client processes of the new output creation 6851 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6852 return id; 6853} 6854 6855status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6856{ 6857 // keep strong reference on the playback thread so that 6858 // it is not destroyed while exit() is executed 6859 sp<PlaybackThread> thread; 6860 { 6861 Mutex::Autolock _l(mLock); 6862 thread = checkPlaybackThread_l(output); 6863 if (thread == NULL) { 6864 return BAD_VALUE; 6865 } 6866 6867 ALOGV("closeOutput() %d", output); 6868 6869 if (thread->type() == ThreadBase::MIXER) { 6870 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6871 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6872 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6873 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6874 } 6875 } 6876 } 6877 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6878 mPlaybackThreads.removeItem(output); 6879 } 6880 thread->exit(); 6881 // The thread entity (active unit of execution) is no longer running here, 6882 // but the ThreadBase container still exists. 6883 6884 if (thread->type() != ThreadBase::DUPLICATING) { 6885 AudioStreamOut *out = thread->clearOutput(); 6886 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6887 // from now on thread->mOutput is NULL 6888 out->hwDev->close_output_stream(out->hwDev, out->stream); 6889 delete out; 6890 } 6891 return NO_ERROR; 6892} 6893 6894status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6895{ 6896 Mutex::Autolock _l(mLock); 6897 PlaybackThread *thread = checkPlaybackThread_l(output); 6898 6899 if (thread == NULL) { 6900 return BAD_VALUE; 6901 } 6902 6903 ALOGV("suspendOutput() %d", output); 6904 thread->suspend(); 6905 6906 return NO_ERROR; 6907} 6908 6909status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6910{ 6911 Mutex::Autolock _l(mLock); 6912 PlaybackThread *thread = checkPlaybackThread_l(output); 6913 6914 if (thread == NULL) { 6915 return BAD_VALUE; 6916 } 6917 6918 ALOGV("restoreOutput() %d", output); 6919 6920 thread->restore(); 6921 6922 return NO_ERROR; 6923} 6924 6925audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6926 audio_devices_t *pDevices, 6927 uint32_t *pSamplingRate, 6928 audio_format_t *pFormat, 6929 uint32_t *pChannelMask) 6930{ 6931 status_t status; 6932 RecordThread *thread = NULL; 6933 struct audio_config config = { 6934 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6935 channel_mask: pChannelMask ? *pChannelMask : 0, 6936 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6937 }; 6938 uint32_t reqSamplingRate = config.sample_rate; 6939 audio_format_t reqFormat = config.format; 6940 audio_channel_mask_t reqChannels = config.channel_mask; 6941 audio_stream_in_t *inStream = NULL; 6942 audio_hw_device_t *inHwDev; 6943 6944 if (pDevices == NULL || *pDevices == 0) { 6945 return 0; 6946 } 6947 6948 Mutex::Autolock _l(mLock); 6949 6950 inHwDev = findSuitableHwDev_l(module, *pDevices); 6951 if (inHwDev == NULL) 6952 return 0; 6953 6954 audio_io_handle_t id = nextUniqueId(); 6955 6956 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6957 &inStream); 6958 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6959 inStream, 6960 config.sample_rate, 6961 config.format, 6962 config.channel_mask, 6963 status); 6964 6965 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6966 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6967 // or stereo to mono conversions on 16 bit PCM inputs. 6968 if (status == BAD_VALUE && 6969 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6970 (config.sample_rate <= 2 * reqSamplingRate) && 6971 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6972 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6973 inStream = NULL; 6974 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6975 } 6976 6977 if (status == NO_ERROR && inStream != NULL) { 6978 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6979 6980 // Start record thread 6981 // RecorThread require both input and output device indication to forward to audio 6982 // pre processing modules 6983 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6984 thread = new RecordThread(this, 6985 input, 6986 reqSamplingRate, 6987 reqChannels, 6988 id, 6989 device); 6990 mRecordThreads.add(id, thread); 6991 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6992 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6993 if (pFormat != NULL) *pFormat = config.format; 6994 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6995 6996 input->stream->common.standby(&input->stream->common); 6997 6998 // notify client processes of the new input creation 6999 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7000 return id; 7001 } 7002 7003 return 0; 7004} 7005 7006status_t AudioFlinger::closeInput(audio_io_handle_t input) 7007{ 7008 // keep strong reference on the record thread so that 7009 // it is not destroyed while exit() is executed 7010 sp<RecordThread> thread; 7011 { 7012 Mutex::Autolock _l(mLock); 7013 thread = checkRecordThread_l(input); 7014 if (thread == NULL) { 7015 return BAD_VALUE; 7016 } 7017 7018 ALOGV("closeInput() %d", input); 7019 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7020 mRecordThreads.removeItem(input); 7021 } 7022 thread->exit(); 7023 // The thread entity (active unit of execution) is no longer running here, 7024 // but the ThreadBase container still exists. 7025 7026 AudioStreamIn *in = thread->clearInput(); 7027 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7028 // from now on thread->mInput is NULL 7029 in->hwDev->close_input_stream(in->hwDev, in->stream); 7030 delete in; 7031 7032 return NO_ERROR; 7033} 7034 7035status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7036{ 7037 Mutex::Autolock _l(mLock); 7038 MixerThread *dstThread = checkMixerThread_l(output); 7039 if (dstThread == NULL) { 7040 ALOGW("setStreamOutput() bad output id %d", output); 7041 return BAD_VALUE; 7042 } 7043 7044 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7045 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7046 7047 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7048 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7049 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7050 MixerThread *srcThread = (MixerThread *)thread; 7051 srcThread->invalidateTracks(stream); 7052 } 7053 } 7054 7055 return NO_ERROR; 7056} 7057 7058 7059int AudioFlinger::newAudioSessionId() 7060{ 7061 return nextUniqueId(); 7062} 7063 7064void AudioFlinger::acquireAudioSessionId(int audioSession) 7065{ 7066 Mutex::Autolock _l(mLock); 7067 pid_t caller = IPCThreadState::self()->getCallingPid(); 7068 ALOGV("acquiring %d from %d", audioSession, caller); 7069 size_t num = mAudioSessionRefs.size(); 7070 for (size_t i = 0; i< num; i++) { 7071 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7072 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7073 ref->mCnt++; 7074 ALOGV(" incremented refcount to %d", ref->mCnt); 7075 return; 7076 } 7077 } 7078 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7079 ALOGV(" added new entry for %d", audioSession); 7080} 7081 7082void AudioFlinger::releaseAudioSessionId(int audioSession) 7083{ 7084 Mutex::Autolock _l(mLock); 7085 pid_t caller = IPCThreadState::self()->getCallingPid(); 7086 ALOGV("releasing %d from %d", audioSession, caller); 7087 size_t num = mAudioSessionRefs.size(); 7088 for (size_t i = 0; i< num; i++) { 7089 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7090 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7091 ref->mCnt--; 7092 ALOGV(" decremented refcount to %d", ref->mCnt); 7093 if (ref->mCnt == 0) { 7094 mAudioSessionRefs.removeAt(i); 7095 delete ref; 7096 purgeStaleEffects_l(); 7097 } 7098 return; 7099 } 7100 } 7101 ALOGW("session id %d not found for pid %d", audioSession, caller); 7102} 7103 7104void AudioFlinger::purgeStaleEffects_l() { 7105 7106 ALOGV("purging stale effects"); 7107 7108 Vector< sp<EffectChain> > chains; 7109 7110 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7111 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7112 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7113 sp<EffectChain> ec = t->mEffectChains[j]; 7114 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7115 chains.push(ec); 7116 } 7117 } 7118 } 7119 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7120 sp<RecordThread> t = mRecordThreads.valueAt(i); 7121 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7122 sp<EffectChain> ec = t->mEffectChains[j]; 7123 chains.push(ec); 7124 } 7125 } 7126 7127 for (size_t i = 0; i < chains.size(); i++) { 7128 sp<EffectChain> ec = chains[i]; 7129 int sessionid = ec->sessionId(); 7130 sp<ThreadBase> t = ec->mThread.promote(); 7131 if (t == 0) { 7132 continue; 7133 } 7134 size_t numsessionrefs = mAudioSessionRefs.size(); 7135 bool found = false; 7136 for (size_t k = 0; k < numsessionrefs; k++) { 7137 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7138 if (ref->mSessionid == sessionid) { 7139 ALOGV(" session %d still exists for %d with %d refs", 7140 sessionid, ref->mPid, ref->mCnt); 7141 found = true; 7142 break; 7143 } 7144 } 7145 if (!found) { 7146 // remove all effects from the chain 7147 while (ec->mEffects.size()) { 7148 sp<EffectModule> effect = ec->mEffects[0]; 7149 effect->unPin(); 7150 Mutex::Autolock _l (t->mLock); 7151 t->removeEffect_l(effect); 7152 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7153 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7154 if (handle != 0) { 7155 handle->mEffect.clear(); 7156 if (handle->mHasControl && handle->mEnabled) { 7157 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7158 } 7159 } 7160 } 7161 AudioSystem::unregisterEffect(effect->id()); 7162 } 7163 } 7164 } 7165 return; 7166} 7167 7168// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7169AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7170{ 7171 return mPlaybackThreads.valueFor(output).get(); 7172} 7173 7174// checkMixerThread_l() must be called with AudioFlinger::mLock held 7175AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7176{ 7177 PlaybackThread *thread = checkPlaybackThread_l(output); 7178 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7179} 7180 7181// checkRecordThread_l() must be called with AudioFlinger::mLock held 7182AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7183{ 7184 return mRecordThreads.valueFor(input).get(); 7185} 7186 7187uint32_t AudioFlinger::nextUniqueId() 7188{ 7189 return android_atomic_inc(&mNextUniqueId); 7190} 7191 7192AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7193{ 7194 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7195 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7196 AudioStreamOut *output = thread->getOutput(); 7197 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7198 return thread; 7199 } 7200 } 7201 return NULL; 7202} 7203 7204uint32_t AudioFlinger::primaryOutputDevice_l() const 7205{ 7206 PlaybackThread *thread = primaryPlaybackThread_l(); 7207 7208 if (thread == NULL) { 7209 return 0; 7210 } 7211 7212 return thread->device(); 7213} 7214 7215sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7216 int triggerSession, 7217 int listenerSession, 7218 sync_event_callback_t callBack, 7219 void *cookie) 7220{ 7221 Mutex::Autolock _l(mLock); 7222 7223 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7224 status_t playStatus = NAME_NOT_FOUND; 7225 status_t recStatus = NAME_NOT_FOUND; 7226 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7227 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7228 if (playStatus == NO_ERROR) { 7229 return event; 7230 } 7231 } 7232 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7233 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7234 if (recStatus == NO_ERROR) { 7235 return event; 7236 } 7237 } 7238 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7239 mPendingSyncEvents.add(event); 7240 } else { 7241 ALOGV("createSyncEvent() invalid event %d", event->type()); 7242 event.clear(); 7243 } 7244 return event; 7245} 7246 7247// ---------------------------------------------------------------------------- 7248// Effect management 7249// ---------------------------------------------------------------------------- 7250 7251 7252status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7253{ 7254 Mutex::Autolock _l(mLock); 7255 return EffectQueryNumberEffects(numEffects); 7256} 7257 7258status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7259{ 7260 Mutex::Autolock _l(mLock); 7261 return EffectQueryEffect(index, descriptor); 7262} 7263 7264status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7265 effect_descriptor_t *descriptor) const 7266{ 7267 Mutex::Autolock _l(mLock); 7268 return EffectGetDescriptor(pUuid, descriptor); 7269} 7270 7271 7272sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7273 effect_descriptor_t *pDesc, 7274 const sp<IEffectClient>& effectClient, 7275 int32_t priority, 7276 audio_io_handle_t io, 7277 int sessionId, 7278 status_t *status, 7279 int *id, 7280 int *enabled) 7281{ 7282 status_t lStatus = NO_ERROR; 7283 sp<EffectHandle> handle; 7284 effect_descriptor_t desc; 7285 7286 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7287 pid, effectClient.get(), priority, sessionId, io); 7288 7289 if (pDesc == NULL) { 7290 lStatus = BAD_VALUE; 7291 goto Exit; 7292 } 7293 7294 // check audio settings permission for global effects 7295 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7296 lStatus = PERMISSION_DENIED; 7297 goto Exit; 7298 } 7299 7300 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7301 // that can only be created by audio policy manager (running in same process) 7302 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7303 lStatus = PERMISSION_DENIED; 7304 goto Exit; 7305 } 7306 7307 if (io == 0) { 7308 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7309 // output must be specified by AudioPolicyManager when using session 7310 // AUDIO_SESSION_OUTPUT_STAGE 7311 lStatus = BAD_VALUE; 7312 goto Exit; 7313 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7314 // if the output returned by getOutputForEffect() is removed before we lock the 7315 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7316 // and we will exit safely 7317 io = AudioSystem::getOutputForEffect(&desc); 7318 } 7319 } 7320 7321 { 7322 Mutex::Autolock _l(mLock); 7323 7324 7325 if (!EffectIsNullUuid(&pDesc->uuid)) { 7326 // if uuid is specified, request effect descriptor 7327 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7328 if (lStatus < 0) { 7329 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7330 goto Exit; 7331 } 7332 } else { 7333 // if uuid is not specified, look for an available implementation 7334 // of the required type in effect factory 7335 if (EffectIsNullUuid(&pDesc->type)) { 7336 ALOGW("createEffect() no effect type"); 7337 lStatus = BAD_VALUE; 7338 goto Exit; 7339 } 7340 uint32_t numEffects = 0; 7341 effect_descriptor_t d; 7342 d.flags = 0; // prevent compiler warning 7343 bool found = false; 7344 7345 lStatus = EffectQueryNumberEffects(&numEffects); 7346 if (lStatus < 0) { 7347 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7348 goto Exit; 7349 } 7350 for (uint32_t i = 0; i < numEffects; i++) { 7351 lStatus = EffectQueryEffect(i, &desc); 7352 if (lStatus < 0) { 7353 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7354 continue; 7355 } 7356 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7357 // If matching type found save effect descriptor. If the session is 7358 // 0 and the effect is not auxiliary, continue enumeration in case 7359 // an auxiliary version of this effect type is available 7360 found = true; 7361 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7362 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7363 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7364 break; 7365 } 7366 } 7367 } 7368 if (!found) { 7369 lStatus = BAD_VALUE; 7370 ALOGW("createEffect() effect not found"); 7371 goto Exit; 7372 } 7373 // For same effect type, chose auxiliary version over insert version if 7374 // connect to output mix (Compliance to OpenSL ES) 7375 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7376 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7377 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7378 } 7379 } 7380 7381 // Do not allow auxiliary effects on a session different from 0 (output mix) 7382 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7383 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7384 lStatus = INVALID_OPERATION; 7385 goto Exit; 7386 } 7387 7388 // check recording permission for visualizer 7389 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7390 !recordingAllowed()) { 7391 lStatus = PERMISSION_DENIED; 7392 goto Exit; 7393 } 7394 7395 // return effect descriptor 7396 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7397 7398 // If output is not specified try to find a matching audio session ID in one of the 7399 // output threads. 7400 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7401 // because of code checking output when entering the function. 7402 // Note: io is never 0 when creating an effect on an input 7403 if (io == 0) { 7404 // look for the thread where the specified audio session is present 7405 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7406 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7407 io = mPlaybackThreads.keyAt(i); 7408 break; 7409 } 7410 } 7411 if (io == 0) { 7412 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7413 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7414 io = mRecordThreads.keyAt(i); 7415 break; 7416 } 7417 } 7418 } 7419 // If no output thread contains the requested session ID, default to 7420 // first output. The effect chain will be moved to the correct output 7421 // thread when a track with the same session ID is created 7422 if (io == 0 && mPlaybackThreads.size()) { 7423 io = mPlaybackThreads.keyAt(0); 7424 } 7425 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7426 } 7427 ThreadBase *thread = checkRecordThread_l(io); 7428 if (thread == NULL) { 7429 thread = checkPlaybackThread_l(io); 7430 if (thread == NULL) { 7431 ALOGE("createEffect() unknown output thread"); 7432 lStatus = BAD_VALUE; 7433 goto Exit; 7434 } 7435 } 7436 7437 sp<Client> client = registerPid_l(pid); 7438 7439 // create effect on selected output thread 7440 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7441 &desc, enabled, &lStatus); 7442 if (handle != 0 && id != NULL) { 7443 *id = handle->id(); 7444 } 7445 } 7446 7447Exit: 7448 if (status != NULL) { 7449 *status = lStatus; 7450 } 7451 return handle; 7452} 7453 7454status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7455 audio_io_handle_t dstOutput) 7456{ 7457 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7458 sessionId, srcOutput, dstOutput); 7459 Mutex::Autolock _l(mLock); 7460 if (srcOutput == dstOutput) { 7461 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7462 return NO_ERROR; 7463 } 7464 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7465 if (srcThread == NULL) { 7466 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7467 return BAD_VALUE; 7468 } 7469 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7470 if (dstThread == NULL) { 7471 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7472 return BAD_VALUE; 7473 } 7474 7475 Mutex::Autolock _dl(dstThread->mLock); 7476 Mutex::Autolock _sl(srcThread->mLock); 7477 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7478 7479 return NO_ERROR; 7480} 7481 7482// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7483status_t AudioFlinger::moveEffectChain_l(int sessionId, 7484 AudioFlinger::PlaybackThread *srcThread, 7485 AudioFlinger::PlaybackThread *dstThread, 7486 bool reRegister) 7487{ 7488 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7489 sessionId, srcThread, dstThread); 7490 7491 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7492 if (chain == 0) { 7493 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7494 sessionId, srcThread); 7495 return INVALID_OPERATION; 7496 } 7497 7498 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7499 // so that a new chain is created with correct parameters when first effect is added. This is 7500 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7501 // removed. 7502 srcThread->removeEffectChain_l(chain); 7503 7504 // transfer all effects one by one so that new effect chain is created on new thread with 7505 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7506 audio_io_handle_t dstOutput = dstThread->id(); 7507 sp<EffectChain> dstChain; 7508 uint32_t strategy = 0; // prevent compiler warning 7509 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7510 while (effect != 0) { 7511 srcThread->removeEffect_l(effect); 7512 dstThread->addEffect_l(effect); 7513 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7514 if (effect->state() == EffectModule::ACTIVE || 7515 effect->state() == EffectModule::STOPPING) { 7516 effect->start(); 7517 } 7518 // if the move request is not received from audio policy manager, the effect must be 7519 // re-registered with the new strategy and output 7520 if (dstChain == 0) { 7521 dstChain = effect->chain().promote(); 7522 if (dstChain == 0) { 7523 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7524 srcThread->addEffect_l(effect); 7525 return NO_INIT; 7526 } 7527 strategy = dstChain->strategy(); 7528 } 7529 if (reRegister) { 7530 AudioSystem::unregisterEffect(effect->id()); 7531 AudioSystem::registerEffect(&effect->desc(), 7532 dstOutput, 7533 strategy, 7534 sessionId, 7535 effect->id()); 7536 } 7537 effect = chain->getEffectFromId_l(0); 7538 } 7539 7540 return NO_ERROR; 7541} 7542 7543 7544// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7545sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7546 const sp<AudioFlinger::Client>& client, 7547 const sp<IEffectClient>& effectClient, 7548 int32_t priority, 7549 int sessionId, 7550 effect_descriptor_t *desc, 7551 int *enabled, 7552 status_t *status 7553 ) 7554{ 7555 sp<EffectModule> effect; 7556 sp<EffectHandle> handle; 7557 status_t lStatus; 7558 sp<EffectChain> chain; 7559 bool chainCreated = false; 7560 bool effectCreated = false; 7561 bool effectRegistered = false; 7562 7563 lStatus = initCheck(); 7564 if (lStatus != NO_ERROR) { 7565 ALOGW("createEffect_l() Audio driver not initialized."); 7566 goto Exit; 7567 } 7568 7569 // Do not allow effects with session ID 0 on direct output or duplicating threads 7570 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7571 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7572 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7573 desc->name, sessionId); 7574 lStatus = BAD_VALUE; 7575 goto Exit; 7576 } 7577 // Only Pre processor effects are allowed on input threads and only on input threads 7578 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7579 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7580 desc->name, desc->flags, mType); 7581 lStatus = BAD_VALUE; 7582 goto Exit; 7583 } 7584 7585 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7586 7587 { // scope for mLock 7588 Mutex::Autolock _l(mLock); 7589 7590 // check for existing effect chain with the requested audio session 7591 chain = getEffectChain_l(sessionId); 7592 if (chain == 0) { 7593 // create a new chain for this session 7594 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7595 chain = new EffectChain(this, sessionId); 7596 addEffectChain_l(chain); 7597 chain->setStrategy(getStrategyForSession_l(sessionId)); 7598 chainCreated = true; 7599 } else { 7600 effect = chain->getEffectFromDesc_l(desc); 7601 } 7602 7603 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7604 7605 if (effect == 0) { 7606 int id = mAudioFlinger->nextUniqueId(); 7607 // Check CPU and memory usage 7608 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7609 if (lStatus != NO_ERROR) { 7610 goto Exit; 7611 } 7612 effectRegistered = true; 7613 // create a new effect module if none present in the chain 7614 effect = new EffectModule(this, chain, desc, id, sessionId); 7615 lStatus = effect->status(); 7616 if (lStatus != NO_ERROR) { 7617 goto Exit; 7618 } 7619 lStatus = chain->addEffect_l(effect); 7620 if (lStatus != NO_ERROR) { 7621 goto Exit; 7622 } 7623 effectCreated = true; 7624 7625 effect->setDevice(mDevice); 7626 effect->setMode(mAudioFlinger->getMode()); 7627 } 7628 // create effect handle and connect it to effect module 7629 handle = new EffectHandle(effect, client, effectClient, priority); 7630 lStatus = effect->addHandle(handle); 7631 if (enabled != NULL) { 7632 *enabled = (int)effect->isEnabled(); 7633 } 7634 } 7635 7636Exit: 7637 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7638 Mutex::Autolock _l(mLock); 7639 if (effectCreated) { 7640 chain->removeEffect_l(effect); 7641 } 7642 if (effectRegistered) { 7643 AudioSystem::unregisterEffect(effect->id()); 7644 } 7645 if (chainCreated) { 7646 removeEffectChain_l(chain); 7647 } 7648 handle.clear(); 7649 } 7650 7651 if (status != NULL) { 7652 *status = lStatus; 7653 } 7654 return handle; 7655} 7656 7657sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7658{ 7659 sp<EffectChain> chain = getEffectChain_l(sessionId); 7660 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7661} 7662 7663// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7664// PlaybackThread::mLock held 7665status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7666{ 7667 // check for existing effect chain with the requested audio session 7668 int sessionId = effect->sessionId(); 7669 sp<EffectChain> chain = getEffectChain_l(sessionId); 7670 bool chainCreated = false; 7671 7672 if (chain == 0) { 7673 // create a new chain for this session 7674 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7675 chain = new EffectChain(this, sessionId); 7676 addEffectChain_l(chain); 7677 chain->setStrategy(getStrategyForSession_l(sessionId)); 7678 chainCreated = true; 7679 } 7680 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7681 7682 if (chain->getEffectFromId_l(effect->id()) != 0) { 7683 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7684 this, effect->desc().name, chain.get()); 7685 return BAD_VALUE; 7686 } 7687 7688 status_t status = chain->addEffect_l(effect); 7689 if (status != NO_ERROR) { 7690 if (chainCreated) { 7691 removeEffectChain_l(chain); 7692 } 7693 return status; 7694 } 7695 7696 effect->setDevice(mDevice); 7697 effect->setMode(mAudioFlinger->getMode()); 7698 return NO_ERROR; 7699} 7700 7701void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7702 7703 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7704 effect_descriptor_t desc = effect->desc(); 7705 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7706 detachAuxEffect_l(effect->id()); 7707 } 7708 7709 sp<EffectChain> chain = effect->chain().promote(); 7710 if (chain != 0) { 7711 // remove effect chain if removing last effect 7712 if (chain->removeEffect_l(effect) == 0) { 7713 removeEffectChain_l(chain); 7714 } 7715 } else { 7716 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7717 } 7718} 7719 7720void AudioFlinger::ThreadBase::lockEffectChains_l( 7721 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7722{ 7723 effectChains = mEffectChains; 7724 for (size_t i = 0; i < mEffectChains.size(); i++) { 7725 mEffectChains[i]->lock(); 7726 } 7727} 7728 7729void AudioFlinger::ThreadBase::unlockEffectChains( 7730 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7731{ 7732 for (size_t i = 0; i < effectChains.size(); i++) { 7733 effectChains[i]->unlock(); 7734 } 7735} 7736 7737sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7738{ 7739 Mutex::Autolock _l(mLock); 7740 return getEffectChain_l(sessionId); 7741} 7742 7743sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7744{ 7745 size_t size = mEffectChains.size(); 7746 for (size_t i = 0; i < size; i++) { 7747 if (mEffectChains[i]->sessionId() == sessionId) { 7748 return mEffectChains[i]; 7749 } 7750 } 7751 return 0; 7752} 7753 7754void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7755{ 7756 Mutex::Autolock _l(mLock); 7757 size_t size = mEffectChains.size(); 7758 for (size_t i = 0; i < size; i++) { 7759 mEffectChains[i]->setMode_l(mode); 7760 } 7761} 7762 7763void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7764 const wp<EffectHandle>& handle, 7765 bool unpinIfLast) { 7766 7767 Mutex::Autolock _l(mLock); 7768 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7769 // delete the effect module if removing last handle on it 7770 if (effect->removeHandle(handle) == 0) { 7771 if (!effect->isPinned() || unpinIfLast) { 7772 removeEffect_l(effect); 7773 AudioSystem::unregisterEffect(effect->id()); 7774 } 7775 } 7776} 7777 7778status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7779{ 7780 int session = chain->sessionId(); 7781 int16_t *buffer = mMixBuffer; 7782 bool ownsBuffer = false; 7783 7784 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7785 if (session > 0) { 7786 // Only one effect chain can be present in direct output thread and it uses 7787 // the mix buffer as input 7788 if (mType != DIRECT) { 7789 size_t numSamples = mNormalFrameCount * mChannelCount; 7790 buffer = new int16_t[numSamples]; 7791 memset(buffer, 0, numSamples * sizeof(int16_t)); 7792 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7793 ownsBuffer = true; 7794 } 7795 7796 // Attach all tracks with same session ID to this chain. 7797 for (size_t i = 0; i < mTracks.size(); ++i) { 7798 sp<Track> track = mTracks[i]; 7799 if (session == track->sessionId()) { 7800 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7801 track->setMainBuffer(buffer); 7802 chain->incTrackCnt(); 7803 } 7804 } 7805 7806 // indicate all active tracks in the chain 7807 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7808 sp<Track> track = mActiveTracks[i].promote(); 7809 if (track == 0) continue; 7810 if (session == track->sessionId()) { 7811 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7812 chain->incActiveTrackCnt(); 7813 } 7814 } 7815 } 7816 7817 chain->setInBuffer(buffer, ownsBuffer); 7818 chain->setOutBuffer(mMixBuffer); 7819 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7820 // chains list in order to be processed last as it contains output stage effects 7821 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7822 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7823 // after track specific effects and before output stage 7824 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7825 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7826 // Effect chain for other sessions are inserted at beginning of effect 7827 // chains list to be processed before output mix effects. Relative order between other 7828 // sessions is not important 7829 size_t size = mEffectChains.size(); 7830 size_t i = 0; 7831 for (i = 0; i < size; i++) { 7832 if (mEffectChains[i]->sessionId() < session) break; 7833 } 7834 mEffectChains.insertAt(chain, i); 7835 checkSuspendOnAddEffectChain_l(chain); 7836 7837 return NO_ERROR; 7838} 7839 7840size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7841{ 7842 int session = chain->sessionId(); 7843 7844 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7845 7846 for (size_t i = 0; i < mEffectChains.size(); i++) { 7847 if (chain == mEffectChains[i]) { 7848 mEffectChains.removeAt(i); 7849 // detach all active tracks from the chain 7850 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7851 sp<Track> track = mActiveTracks[i].promote(); 7852 if (track == 0) continue; 7853 if (session == track->sessionId()) { 7854 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7855 chain.get(), session); 7856 chain->decActiveTrackCnt(); 7857 } 7858 } 7859 7860 // detach all tracks with same session ID from this chain 7861 for (size_t i = 0; i < mTracks.size(); ++i) { 7862 sp<Track> track = mTracks[i]; 7863 if (session == track->sessionId()) { 7864 track->setMainBuffer(mMixBuffer); 7865 chain->decTrackCnt(); 7866 } 7867 } 7868 break; 7869 } 7870 } 7871 return mEffectChains.size(); 7872} 7873 7874status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7875 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7876{ 7877 Mutex::Autolock _l(mLock); 7878 return attachAuxEffect_l(track, EffectId); 7879} 7880 7881status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7882 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7883{ 7884 status_t status = NO_ERROR; 7885 7886 if (EffectId == 0) { 7887 track->setAuxBuffer(0, NULL); 7888 } else { 7889 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7890 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7891 if (effect != 0) { 7892 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7893 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7894 } else { 7895 status = INVALID_OPERATION; 7896 } 7897 } else { 7898 status = BAD_VALUE; 7899 } 7900 } 7901 return status; 7902} 7903 7904void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7905{ 7906 for (size_t i = 0; i < mTracks.size(); ++i) { 7907 sp<Track> track = mTracks[i]; 7908 if (track->auxEffectId() == effectId) { 7909 attachAuxEffect_l(track, 0); 7910 } 7911 } 7912} 7913 7914status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7915{ 7916 // only one chain per input thread 7917 if (mEffectChains.size() != 0) { 7918 return INVALID_OPERATION; 7919 } 7920 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7921 7922 chain->setInBuffer(NULL); 7923 chain->setOutBuffer(NULL); 7924 7925 checkSuspendOnAddEffectChain_l(chain); 7926 7927 mEffectChains.add(chain); 7928 7929 return NO_ERROR; 7930} 7931 7932size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7933{ 7934 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7935 ALOGW_IF(mEffectChains.size() != 1, 7936 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7937 chain.get(), mEffectChains.size(), this); 7938 if (mEffectChains.size() == 1) { 7939 mEffectChains.removeAt(0); 7940 } 7941 return 0; 7942} 7943 7944// ---------------------------------------------------------------------------- 7945// EffectModule implementation 7946// ---------------------------------------------------------------------------- 7947 7948#undef LOG_TAG 7949#define LOG_TAG "AudioFlinger::EffectModule" 7950 7951AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7952 const wp<AudioFlinger::EffectChain>& chain, 7953 effect_descriptor_t *desc, 7954 int id, 7955 int sessionId) 7956 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7957 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7958{ 7959 ALOGV("Constructor %p", this); 7960 int lStatus; 7961 if (thread == NULL) { 7962 return; 7963 } 7964 7965 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7966 7967 // create effect engine from effect factory 7968 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7969 7970 if (mStatus != NO_ERROR) { 7971 return; 7972 } 7973 lStatus = init(); 7974 if (lStatus < 0) { 7975 mStatus = lStatus; 7976 goto Error; 7977 } 7978 7979 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7980 mPinned = true; 7981 } 7982 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7983 return; 7984Error: 7985 EffectRelease(mEffectInterface); 7986 mEffectInterface = NULL; 7987 ALOGV("Constructor Error %d", mStatus); 7988} 7989 7990AudioFlinger::EffectModule::~EffectModule() 7991{ 7992 ALOGV("Destructor %p", this); 7993 if (mEffectInterface != NULL) { 7994 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7995 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7996 sp<ThreadBase> thread = mThread.promote(); 7997 if (thread != 0) { 7998 audio_stream_t *stream = thread->stream(); 7999 if (stream != NULL) { 8000 stream->remove_audio_effect(stream, mEffectInterface); 8001 } 8002 } 8003 } 8004 // release effect engine 8005 EffectRelease(mEffectInterface); 8006 } 8007} 8008 8009status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 8010{ 8011 status_t status; 8012 8013 Mutex::Autolock _l(mLock); 8014 int priority = handle->priority(); 8015 size_t size = mHandles.size(); 8016 sp<EffectHandle> h; 8017 size_t i; 8018 for (i = 0; i < size; i++) { 8019 h = mHandles[i].promote(); 8020 if (h == 0) continue; 8021 if (h->priority() <= priority) break; 8022 } 8023 // if inserted in first place, move effect control from previous owner to this handle 8024 if (i == 0) { 8025 bool enabled = false; 8026 if (h != 0) { 8027 enabled = h->enabled(); 8028 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8029 } 8030 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8031 status = NO_ERROR; 8032 } else { 8033 status = ALREADY_EXISTS; 8034 } 8035 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8036 mHandles.insertAt(handle, i); 8037 return status; 8038} 8039 8040size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8041{ 8042 Mutex::Autolock _l(mLock); 8043 size_t size = mHandles.size(); 8044 size_t i; 8045 for (i = 0; i < size; i++) { 8046 if (mHandles[i] == handle) break; 8047 } 8048 if (i == size) { 8049 return size; 8050 } 8051 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8052 8053 bool enabled = false; 8054 EffectHandle *hdl = handle.unsafe_get(); 8055 if (hdl != NULL) { 8056 ALOGV("removeHandle() unsafe_get OK"); 8057 enabled = hdl->enabled(); 8058 } 8059 mHandles.removeAt(i); 8060 size = mHandles.size(); 8061 // if removed from first place, move effect control from this handle to next in line 8062 if (i == 0 && size != 0) { 8063 sp<EffectHandle> h = mHandles[0].promote(); 8064 if (h != 0) { 8065 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8066 } 8067 } 8068 8069 // Prevent calls to process() and other functions on effect interface from now on. 8070 // The effect engine will be released by the destructor when the last strong reference on 8071 // this object is released which can happen after next process is called. 8072 if (size == 0 && !mPinned) { 8073 mState = DESTROYED; 8074 } 8075 8076 return size; 8077} 8078 8079sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8080{ 8081 Mutex::Autolock _l(mLock); 8082 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8083} 8084 8085void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8086{ 8087 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8088 // keep a strong reference on this EffectModule to avoid calling the 8089 // destructor before we exit 8090 sp<EffectModule> keep(this); 8091 { 8092 sp<ThreadBase> thread = mThread.promote(); 8093 if (thread != 0) { 8094 thread->disconnectEffect(keep, handle, unpinIfLast); 8095 } 8096 } 8097} 8098 8099void AudioFlinger::EffectModule::updateState() { 8100 Mutex::Autolock _l(mLock); 8101 8102 switch (mState) { 8103 case RESTART: 8104 reset_l(); 8105 // FALL THROUGH 8106 8107 case STARTING: 8108 // clear auxiliary effect input buffer for next accumulation 8109 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8110 memset(mConfig.inputCfg.buffer.raw, 8111 0, 8112 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8113 } 8114 start_l(); 8115 mState = ACTIVE; 8116 break; 8117 case STOPPING: 8118 stop_l(); 8119 mDisableWaitCnt = mMaxDisableWaitCnt; 8120 mState = STOPPED; 8121 break; 8122 case STOPPED: 8123 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8124 // turn off sequence. 8125 if (--mDisableWaitCnt == 0) { 8126 reset_l(); 8127 mState = IDLE; 8128 } 8129 break; 8130 default: //IDLE , ACTIVE, DESTROYED 8131 break; 8132 } 8133} 8134 8135void AudioFlinger::EffectModule::process() 8136{ 8137 Mutex::Autolock _l(mLock); 8138 8139 if (mState == DESTROYED || mEffectInterface == NULL || 8140 mConfig.inputCfg.buffer.raw == NULL || 8141 mConfig.outputCfg.buffer.raw == NULL) { 8142 return; 8143 } 8144 8145 if (isProcessEnabled()) { 8146 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8147 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8148 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8149 mConfig.inputCfg.buffer.s32, 8150 mConfig.inputCfg.buffer.frameCount/2); 8151 } 8152 8153 // do the actual processing in the effect engine 8154 int ret = (*mEffectInterface)->process(mEffectInterface, 8155 &mConfig.inputCfg.buffer, 8156 &mConfig.outputCfg.buffer); 8157 8158 // force transition to IDLE state when engine is ready 8159 if (mState == STOPPED && ret == -ENODATA) { 8160 mDisableWaitCnt = 1; 8161 } 8162 8163 // clear auxiliary effect input buffer for next accumulation 8164 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8165 memset(mConfig.inputCfg.buffer.raw, 0, 8166 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8167 } 8168 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8169 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8170 // If an insert effect is idle and input buffer is different from output buffer, 8171 // accumulate input onto output 8172 sp<EffectChain> chain = mChain.promote(); 8173 if (chain != 0 && chain->activeTrackCnt() != 0) { 8174 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8175 int16_t *in = mConfig.inputCfg.buffer.s16; 8176 int16_t *out = mConfig.outputCfg.buffer.s16; 8177 for (size_t i = 0; i < frameCnt; i++) { 8178 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8179 } 8180 } 8181 } 8182} 8183 8184void AudioFlinger::EffectModule::reset_l() 8185{ 8186 if (mEffectInterface == NULL) { 8187 return; 8188 } 8189 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8190} 8191 8192status_t AudioFlinger::EffectModule::configure() 8193{ 8194 uint32_t channels; 8195 if (mEffectInterface == NULL) { 8196 return NO_INIT; 8197 } 8198 8199 sp<ThreadBase> thread = mThread.promote(); 8200 if (thread == 0) { 8201 return DEAD_OBJECT; 8202 } 8203 8204 // TODO: handle configuration of effects replacing track process 8205 if (thread->channelCount() == 1) { 8206 channels = AUDIO_CHANNEL_OUT_MONO; 8207 } else { 8208 channels = AUDIO_CHANNEL_OUT_STEREO; 8209 } 8210 8211 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8212 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8213 } else { 8214 mConfig.inputCfg.channels = channels; 8215 } 8216 mConfig.outputCfg.channels = channels; 8217 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8218 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8219 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8220 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8221 mConfig.inputCfg.bufferProvider.cookie = NULL; 8222 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8223 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8224 mConfig.outputCfg.bufferProvider.cookie = NULL; 8225 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8226 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8227 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8228 // Insert effect: 8229 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8230 // always overwrites output buffer: input buffer == output buffer 8231 // - in other sessions: 8232 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8233 // other effect: overwrites output buffer: input buffer == output buffer 8234 // Auxiliary effect: 8235 // accumulates in output buffer: input buffer != output buffer 8236 // Therefore: accumulate <=> input buffer != output buffer 8237 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8238 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8239 } else { 8240 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8241 } 8242 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8243 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8244 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8245 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8246 8247 ALOGV("configure() %p thread %p buffer %p framecount %d", 8248 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8249 8250 status_t cmdStatus; 8251 uint32_t size = sizeof(int); 8252 status_t status = (*mEffectInterface)->command(mEffectInterface, 8253 EFFECT_CMD_SET_CONFIG, 8254 sizeof(effect_config_t), 8255 &mConfig, 8256 &size, 8257 &cmdStatus); 8258 if (status == 0) { 8259 status = cmdStatus; 8260 } 8261 8262 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8263 (1000 * mConfig.outputCfg.buffer.frameCount); 8264 8265 return status; 8266} 8267 8268status_t AudioFlinger::EffectModule::init() 8269{ 8270 Mutex::Autolock _l(mLock); 8271 if (mEffectInterface == NULL) { 8272 return NO_INIT; 8273 } 8274 status_t cmdStatus; 8275 uint32_t size = sizeof(status_t); 8276 status_t status = (*mEffectInterface)->command(mEffectInterface, 8277 EFFECT_CMD_INIT, 8278 0, 8279 NULL, 8280 &size, 8281 &cmdStatus); 8282 if (status == 0) { 8283 status = cmdStatus; 8284 } 8285 return status; 8286} 8287 8288status_t AudioFlinger::EffectModule::start() 8289{ 8290 Mutex::Autolock _l(mLock); 8291 return start_l(); 8292} 8293 8294status_t AudioFlinger::EffectModule::start_l() 8295{ 8296 if (mEffectInterface == NULL) { 8297 return NO_INIT; 8298 } 8299 status_t cmdStatus; 8300 uint32_t size = sizeof(status_t); 8301 status_t status = (*mEffectInterface)->command(mEffectInterface, 8302 EFFECT_CMD_ENABLE, 8303 0, 8304 NULL, 8305 &size, 8306 &cmdStatus); 8307 if (status == 0) { 8308 status = cmdStatus; 8309 } 8310 if (status == 0 && 8311 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8312 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8313 sp<ThreadBase> thread = mThread.promote(); 8314 if (thread != 0) { 8315 audio_stream_t *stream = thread->stream(); 8316 if (stream != NULL) { 8317 stream->add_audio_effect(stream, mEffectInterface); 8318 } 8319 } 8320 } 8321 return status; 8322} 8323 8324status_t AudioFlinger::EffectModule::stop() 8325{ 8326 Mutex::Autolock _l(mLock); 8327 return stop_l(); 8328} 8329 8330status_t AudioFlinger::EffectModule::stop_l() 8331{ 8332 if (mEffectInterface == NULL) { 8333 return NO_INIT; 8334 } 8335 status_t cmdStatus; 8336 uint32_t size = sizeof(status_t); 8337 status_t status = (*mEffectInterface)->command(mEffectInterface, 8338 EFFECT_CMD_DISABLE, 8339 0, 8340 NULL, 8341 &size, 8342 &cmdStatus); 8343 if (status == 0) { 8344 status = cmdStatus; 8345 } 8346 if (status == 0 && 8347 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8348 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8349 sp<ThreadBase> thread = mThread.promote(); 8350 if (thread != 0) { 8351 audio_stream_t *stream = thread->stream(); 8352 if (stream != NULL) { 8353 stream->remove_audio_effect(stream, mEffectInterface); 8354 } 8355 } 8356 } 8357 return status; 8358} 8359 8360status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8361 uint32_t cmdSize, 8362 void *pCmdData, 8363 uint32_t *replySize, 8364 void *pReplyData) 8365{ 8366 Mutex::Autolock _l(mLock); 8367// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8368 8369 if (mState == DESTROYED || mEffectInterface == NULL) { 8370 return NO_INIT; 8371 } 8372 status_t status = (*mEffectInterface)->command(mEffectInterface, 8373 cmdCode, 8374 cmdSize, 8375 pCmdData, 8376 replySize, 8377 pReplyData); 8378 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8379 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8380 for (size_t i = 1; i < mHandles.size(); i++) { 8381 sp<EffectHandle> h = mHandles[i].promote(); 8382 if (h != 0) { 8383 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8384 } 8385 } 8386 } 8387 return status; 8388} 8389 8390status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8391{ 8392 8393 Mutex::Autolock _l(mLock); 8394 ALOGV("setEnabled %p enabled %d", this, enabled); 8395 8396 if (enabled != isEnabled()) { 8397 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8398 if (enabled && status != NO_ERROR) { 8399 return status; 8400 } 8401 8402 switch (mState) { 8403 // going from disabled to enabled 8404 case IDLE: 8405 mState = STARTING; 8406 break; 8407 case STOPPED: 8408 mState = RESTART; 8409 break; 8410 case STOPPING: 8411 mState = ACTIVE; 8412 break; 8413 8414 // going from enabled to disabled 8415 case RESTART: 8416 mState = STOPPED; 8417 break; 8418 case STARTING: 8419 mState = IDLE; 8420 break; 8421 case ACTIVE: 8422 mState = STOPPING; 8423 break; 8424 case DESTROYED: 8425 return NO_ERROR; // simply ignore as we are being destroyed 8426 } 8427 for (size_t i = 1; i < mHandles.size(); i++) { 8428 sp<EffectHandle> h = mHandles[i].promote(); 8429 if (h != 0) { 8430 h->setEnabled(enabled); 8431 } 8432 } 8433 } 8434 return NO_ERROR; 8435} 8436 8437bool AudioFlinger::EffectModule::isEnabled() const 8438{ 8439 switch (mState) { 8440 case RESTART: 8441 case STARTING: 8442 case ACTIVE: 8443 return true; 8444 case IDLE: 8445 case STOPPING: 8446 case STOPPED: 8447 case DESTROYED: 8448 default: 8449 return false; 8450 } 8451} 8452 8453bool AudioFlinger::EffectModule::isProcessEnabled() const 8454{ 8455 switch (mState) { 8456 case RESTART: 8457 case ACTIVE: 8458 case STOPPING: 8459 case STOPPED: 8460 return true; 8461 case IDLE: 8462 case STARTING: 8463 case DESTROYED: 8464 default: 8465 return false; 8466 } 8467} 8468 8469status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8470{ 8471 Mutex::Autolock _l(mLock); 8472 status_t status = NO_ERROR; 8473 8474 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8475 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8476 if (isProcessEnabled() && 8477 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8478 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8479 status_t cmdStatus; 8480 uint32_t volume[2]; 8481 uint32_t *pVolume = NULL; 8482 uint32_t size = sizeof(volume); 8483 volume[0] = *left; 8484 volume[1] = *right; 8485 if (controller) { 8486 pVolume = volume; 8487 } 8488 status = (*mEffectInterface)->command(mEffectInterface, 8489 EFFECT_CMD_SET_VOLUME, 8490 size, 8491 volume, 8492 &size, 8493 pVolume); 8494 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8495 *left = volume[0]; 8496 *right = volume[1]; 8497 } 8498 } 8499 return status; 8500} 8501 8502status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8503{ 8504 Mutex::Autolock _l(mLock); 8505 status_t status = NO_ERROR; 8506 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8507 // audio pre processing modules on RecordThread can receive both output and 8508 // input device indication in the same call 8509 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8510 if (dev) { 8511 status_t cmdStatus; 8512 uint32_t size = sizeof(status_t); 8513 8514 status = (*mEffectInterface)->command(mEffectInterface, 8515 EFFECT_CMD_SET_DEVICE, 8516 sizeof(uint32_t), 8517 &dev, 8518 &size, 8519 &cmdStatus); 8520 if (status == NO_ERROR) { 8521 status = cmdStatus; 8522 } 8523 } 8524 dev = device & AUDIO_DEVICE_IN_ALL; 8525 if (dev) { 8526 status_t cmdStatus; 8527 uint32_t size = sizeof(status_t); 8528 8529 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8530 EFFECT_CMD_SET_INPUT_DEVICE, 8531 sizeof(uint32_t), 8532 &dev, 8533 &size, 8534 &cmdStatus); 8535 if (status2 == NO_ERROR) { 8536 status2 = cmdStatus; 8537 } 8538 if (status == NO_ERROR) { 8539 status = status2; 8540 } 8541 } 8542 } 8543 return status; 8544} 8545 8546status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8547{ 8548 Mutex::Autolock _l(mLock); 8549 status_t status = NO_ERROR; 8550 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8551 status_t cmdStatus; 8552 uint32_t size = sizeof(status_t); 8553 status = (*mEffectInterface)->command(mEffectInterface, 8554 EFFECT_CMD_SET_AUDIO_MODE, 8555 sizeof(audio_mode_t), 8556 &mode, 8557 &size, 8558 &cmdStatus); 8559 if (status == NO_ERROR) { 8560 status = cmdStatus; 8561 } 8562 } 8563 return status; 8564} 8565 8566void AudioFlinger::EffectModule::setSuspended(bool suspended) 8567{ 8568 Mutex::Autolock _l(mLock); 8569 mSuspended = suspended; 8570} 8571 8572bool AudioFlinger::EffectModule::suspended() const 8573{ 8574 Mutex::Autolock _l(mLock); 8575 return mSuspended; 8576} 8577 8578status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8579{ 8580 const size_t SIZE = 256; 8581 char buffer[SIZE]; 8582 String8 result; 8583 8584 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8585 result.append(buffer); 8586 8587 bool locked = tryLock(mLock); 8588 // failed to lock - AudioFlinger is probably deadlocked 8589 if (!locked) { 8590 result.append("\t\tCould not lock Fx mutex:\n"); 8591 } 8592 8593 result.append("\t\tSession Status State Engine:\n"); 8594 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8595 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8596 result.append(buffer); 8597 8598 result.append("\t\tDescriptor:\n"); 8599 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8600 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8601 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8602 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8603 result.append(buffer); 8604 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8605 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8606 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8607 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8608 result.append(buffer); 8609 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8610 mDescriptor.apiVersion, 8611 mDescriptor.flags); 8612 result.append(buffer); 8613 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8614 mDescriptor.name); 8615 result.append(buffer); 8616 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8617 mDescriptor.implementor); 8618 result.append(buffer); 8619 8620 result.append("\t\t- Input configuration:\n"); 8621 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8622 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8623 (uint32_t)mConfig.inputCfg.buffer.raw, 8624 mConfig.inputCfg.buffer.frameCount, 8625 mConfig.inputCfg.samplingRate, 8626 mConfig.inputCfg.channels, 8627 mConfig.inputCfg.format); 8628 result.append(buffer); 8629 8630 result.append("\t\t- Output configuration:\n"); 8631 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8632 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8633 (uint32_t)mConfig.outputCfg.buffer.raw, 8634 mConfig.outputCfg.buffer.frameCount, 8635 mConfig.outputCfg.samplingRate, 8636 mConfig.outputCfg.channels, 8637 mConfig.outputCfg.format); 8638 result.append(buffer); 8639 8640 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8641 result.append(buffer); 8642 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8643 for (size_t i = 0; i < mHandles.size(); ++i) { 8644 sp<EffectHandle> handle = mHandles[i].promote(); 8645 if (handle != 0) { 8646 handle->dump(buffer, SIZE); 8647 result.append(buffer); 8648 } 8649 } 8650 8651 result.append("\n"); 8652 8653 write(fd, result.string(), result.length()); 8654 8655 if (locked) { 8656 mLock.unlock(); 8657 } 8658 8659 return NO_ERROR; 8660} 8661 8662// ---------------------------------------------------------------------------- 8663// EffectHandle implementation 8664// ---------------------------------------------------------------------------- 8665 8666#undef LOG_TAG 8667#define LOG_TAG "AudioFlinger::EffectHandle" 8668 8669AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8670 const sp<AudioFlinger::Client>& client, 8671 const sp<IEffectClient>& effectClient, 8672 int32_t priority) 8673 : BnEffect(), 8674 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8675 mPriority(priority), mHasControl(false), mEnabled(false) 8676{ 8677 ALOGV("constructor %p", this); 8678 8679 if (client == 0) { 8680 return; 8681 } 8682 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8683 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8684 if (mCblkMemory != 0) { 8685 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8686 8687 if (mCblk != NULL) { 8688 new(mCblk) effect_param_cblk_t(); 8689 mBuffer = (uint8_t *)mCblk + bufOffset; 8690 } 8691 } else { 8692 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8693 return; 8694 } 8695} 8696 8697AudioFlinger::EffectHandle::~EffectHandle() 8698{ 8699 ALOGV("Destructor %p", this); 8700 disconnect(false); 8701 ALOGV("Destructor DONE %p", this); 8702} 8703 8704status_t AudioFlinger::EffectHandle::enable() 8705{ 8706 ALOGV("enable %p", this); 8707 if (!mHasControl) return INVALID_OPERATION; 8708 if (mEffect == 0) return DEAD_OBJECT; 8709 8710 if (mEnabled) { 8711 return NO_ERROR; 8712 } 8713 8714 mEnabled = true; 8715 8716 sp<ThreadBase> thread = mEffect->thread().promote(); 8717 if (thread != 0) { 8718 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8719 } 8720 8721 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8722 if (mEffect->suspended()) { 8723 return NO_ERROR; 8724 } 8725 8726 status_t status = mEffect->setEnabled(true); 8727 if (status != NO_ERROR) { 8728 if (thread != 0) { 8729 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8730 } 8731 mEnabled = false; 8732 } 8733 return status; 8734} 8735 8736status_t AudioFlinger::EffectHandle::disable() 8737{ 8738 ALOGV("disable %p", this); 8739 if (!mHasControl) return INVALID_OPERATION; 8740 if (mEffect == 0) return DEAD_OBJECT; 8741 8742 if (!mEnabled) { 8743 return NO_ERROR; 8744 } 8745 mEnabled = false; 8746 8747 if (mEffect->suspended()) { 8748 return NO_ERROR; 8749 } 8750 8751 status_t status = mEffect->setEnabled(false); 8752 8753 sp<ThreadBase> thread = mEffect->thread().promote(); 8754 if (thread != 0) { 8755 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8756 } 8757 8758 return status; 8759} 8760 8761void AudioFlinger::EffectHandle::disconnect() 8762{ 8763 disconnect(true); 8764} 8765 8766void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8767{ 8768 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8769 if (mEffect == 0) { 8770 return; 8771 } 8772 mEffect->disconnect(this, unpinIfLast); 8773 8774 if (mHasControl && mEnabled) { 8775 sp<ThreadBase> thread = mEffect->thread().promote(); 8776 if (thread != 0) { 8777 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8778 } 8779 } 8780 8781 // release sp on module => module destructor can be called now 8782 mEffect.clear(); 8783 if (mClient != 0) { 8784 if (mCblk != NULL) { 8785 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8786 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8787 } 8788 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8789 // Client destructor must run with AudioFlinger mutex locked 8790 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8791 mClient.clear(); 8792 } 8793} 8794 8795status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8796 uint32_t cmdSize, 8797 void *pCmdData, 8798 uint32_t *replySize, 8799 void *pReplyData) 8800{ 8801// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8802// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8803 8804 // only get parameter command is permitted for applications not controlling the effect 8805 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8806 return INVALID_OPERATION; 8807 } 8808 if (mEffect == 0) return DEAD_OBJECT; 8809 if (mClient == 0) return INVALID_OPERATION; 8810 8811 // handle commands that are not forwarded transparently to effect engine 8812 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8813 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8814 // no risk to block the whole media server process or mixer threads is we are stuck here 8815 Mutex::Autolock _l(mCblk->lock); 8816 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8817 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8818 mCblk->serverIndex = 0; 8819 mCblk->clientIndex = 0; 8820 return BAD_VALUE; 8821 } 8822 status_t status = NO_ERROR; 8823 while (mCblk->serverIndex < mCblk->clientIndex) { 8824 int reply; 8825 uint32_t rsize = sizeof(int); 8826 int *p = (int *)(mBuffer + mCblk->serverIndex); 8827 int size = *p++; 8828 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8829 ALOGW("command(): invalid parameter block size"); 8830 break; 8831 } 8832 effect_param_t *param = (effect_param_t *)p; 8833 if (param->psize == 0 || param->vsize == 0) { 8834 ALOGW("command(): null parameter or value size"); 8835 mCblk->serverIndex += size; 8836 continue; 8837 } 8838 uint32_t psize = sizeof(effect_param_t) + 8839 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8840 param->vsize; 8841 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8842 psize, 8843 p, 8844 &rsize, 8845 &reply); 8846 // stop at first error encountered 8847 if (ret != NO_ERROR) { 8848 status = ret; 8849 *(int *)pReplyData = reply; 8850 break; 8851 } else if (reply != NO_ERROR) { 8852 *(int *)pReplyData = reply; 8853 break; 8854 } 8855 mCblk->serverIndex += size; 8856 } 8857 mCblk->serverIndex = 0; 8858 mCblk->clientIndex = 0; 8859 return status; 8860 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8861 *(int *)pReplyData = NO_ERROR; 8862 return enable(); 8863 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8864 *(int *)pReplyData = NO_ERROR; 8865 return disable(); 8866 } 8867 8868 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8869} 8870 8871void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8872{ 8873 ALOGV("setControl %p control %d", this, hasControl); 8874 8875 mHasControl = hasControl; 8876 mEnabled = enabled; 8877 8878 if (signal && mEffectClient != 0) { 8879 mEffectClient->controlStatusChanged(hasControl); 8880 } 8881} 8882 8883void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8884 uint32_t cmdSize, 8885 void *pCmdData, 8886 uint32_t replySize, 8887 void *pReplyData) 8888{ 8889 if (mEffectClient != 0) { 8890 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8891 } 8892} 8893 8894 8895 8896void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8897{ 8898 if (mEffectClient != 0) { 8899 mEffectClient->enableStatusChanged(enabled); 8900 } 8901} 8902 8903status_t AudioFlinger::EffectHandle::onTransact( 8904 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8905{ 8906 return BnEffect::onTransact(code, data, reply, flags); 8907} 8908 8909 8910void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8911{ 8912 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8913 8914 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8915 (mClient == 0) ? getpid_cached : mClient->pid(), 8916 mPriority, 8917 mHasControl, 8918 !locked, 8919 mCblk ? mCblk->clientIndex : 0, 8920 mCblk ? mCblk->serverIndex : 0 8921 ); 8922 8923 if (locked) { 8924 mCblk->lock.unlock(); 8925 } 8926} 8927 8928#undef LOG_TAG 8929#define LOG_TAG "AudioFlinger::EffectChain" 8930 8931AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8932 int sessionId) 8933 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8934 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8935 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8936{ 8937 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8938 if (thread == NULL) { 8939 return; 8940 } 8941 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8942 thread->frameCount(); 8943} 8944 8945AudioFlinger::EffectChain::~EffectChain() 8946{ 8947 if (mOwnInBuffer) { 8948 delete mInBuffer; 8949 } 8950 8951} 8952 8953// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8954sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8955{ 8956 size_t size = mEffects.size(); 8957 8958 for (size_t i = 0; i < size; i++) { 8959 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8960 return mEffects[i]; 8961 } 8962 } 8963 return 0; 8964} 8965 8966// getEffectFromId_l() must be called with ThreadBase::mLock held 8967sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8968{ 8969 size_t size = mEffects.size(); 8970 8971 for (size_t i = 0; i < size; i++) { 8972 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8973 if (id == 0 || mEffects[i]->id() == id) { 8974 return mEffects[i]; 8975 } 8976 } 8977 return 0; 8978} 8979 8980// getEffectFromType_l() must be called with ThreadBase::mLock held 8981sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8982 const effect_uuid_t *type) 8983{ 8984 size_t size = mEffects.size(); 8985 8986 for (size_t i = 0; i < size; i++) { 8987 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8988 return mEffects[i]; 8989 } 8990 } 8991 return 0; 8992} 8993 8994void AudioFlinger::EffectChain::clearInputBuffer() 8995{ 8996 Mutex::Autolock _l(mLock); 8997 sp<ThreadBase> thread = mThread.promote(); 8998 if (thread == 0) { 8999 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9000 return; 9001 } 9002 clearInputBuffer_l(thread); 9003} 9004 9005// Must be called with EffectChain::mLock locked 9006void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9007{ 9008 size_t numSamples = thread->frameCount() * thread->channelCount(); 9009 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9010 9011} 9012 9013// Must be called with EffectChain::mLock locked 9014void AudioFlinger::EffectChain::process_l() 9015{ 9016 sp<ThreadBase> thread = mThread.promote(); 9017 if (thread == 0) { 9018 ALOGW("process_l(): cannot promote mixer thread"); 9019 return; 9020 } 9021 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9022 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9023 // always process effects unless no more tracks are on the session and the effect tail 9024 // has been rendered 9025 bool doProcess = true; 9026 if (!isGlobalSession) { 9027 bool tracksOnSession = (trackCnt() != 0); 9028 9029 if (!tracksOnSession && mTailBufferCount == 0) { 9030 doProcess = false; 9031 } 9032 9033 if (activeTrackCnt() == 0) { 9034 // if no track is active and the effect tail has not been rendered, 9035 // the input buffer must be cleared here as the mixer process will not do it 9036 if (tracksOnSession || mTailBufferCount > 0) { 9037 clearInputBuffer_l(thread); 9038 if (mTailBufferCount > 0) { 9039 mTailBufferCount--; 9040 } 9041 } 9042 } 9043 } 9044 9045 size_t size = mEffects.size(); 9046 if (doProcess) { 9047 for (size_t i = 0; i < size; i++) { 9048 mEffects[i]->process(); 9049 } 9050 } 9051 for (size_t i = 0; i < size; i++) { 9052 mEffects[i]->updateState(); 9053 } 9054} 9055 9056// addEffect_l() must be called with PlaybackThread::mLock held 9057status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9058{ 9059 effect_descriptor_t desc = effect->desc(); 9060 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9061 9062 Mutex::Autolock _l(mLock); 9063 effect->setChain(this); 9064 sp<ThreadBase> thread = mThread.promote(); 9065 if (thread == 0) { 9066 return NO_INIT; 9067 } 9068 effect->setThread(thread); 9069 9070 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9071 // Auxiliary effects are inserted at the beginning of mEffects vector as 9072 // they are processed first and accumulated in chain input buffer 9073 mEffects.insertAt(effect, 0); 9074 9075 // the input buffer for auxiliary effect contains mono samples in 9076 // 32 bit format. This is to avoid saturation in AudoMixer 9077 // accumulation stage. Saturation is done in EffectModule::process() before 9078 // calling the process in effect engine 9079 size_t numSamples = thread->frameCount(); 9080 int32_t *buffer = new int32_t[numSamples]; 9081 memset(buffer, 0, numSamples * sizeof(int32_t)); 9082 effect->setInBuffer((int16_t *)buffer); 9083 // auxiliary effects output samples to chain input buffer for further processing 9084 // by insert effects 9085 effect->setOutBuffer(mInBuffer); 9086 } else { 9087 // Insert effects are inserted at the end of mEffects vector as they are processed 9088 // after track and auxiliary effects. 9089 // Insert effect order as a function of indicated preference: 9090 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9091 // another effect is present 9092 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9093 // last effect claiming first position 9094 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9095 // first effect claiming last position 9096 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9097 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9098 // already present 9099 9100 size_t size = mEffects.size(); 9101 size_t idx_insert = size; 9102 ssize_t idx_insert_first = -1; 9103 ssize_t idx_insert_last = -1; 9104 9105 for (size_t i = 0; i < size; i++) { 9106 effect_descriptor_t d = mEffects[i]->desc(); 9107 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9108 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9109 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9110 // check invalid effect chaining combinations 9111 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9112 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9113 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9114 return INVALID_OPERATION; 9115 } 9116 // remember position of first insert effect and by default 9117 // select this as insert position for new effect 9118 if (idx_insert == size) { 9119 idx_insert = i; 9120 } 9121 // remember position of last insert effect claiming 9122 // first position 9123 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9124 idx_insert_first = i; 9125 } 9126 // remember position of first insert effect claiming 9127 // last position 9128 if (iPref == EFFECT_FLAG_INSERT_LAST && 9129 idx_insert_last == -1) { 9130 idx_insert_last = i; 9131 } 9132 } 9133 } 9134 9135 // modify idx_insert from first position if needed 9136 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9137 if (idx_insert_last != -1) { 9138 idx_insert = idx_insert_last; 9139 } else { 9140 idx_insert = size; 9141 } 9142 } else { 9143 if (idx_insert_first != -1) { 9144 idx_insert = idx_insert_first + 1; 9145 } 9146 } 9147 9148 // always read samples from chain input buffer 9149 effect->setInBuffer(mInBuffer); 9150 9151 // if last effect in the chain, output samples to chain 9152 // output buffer, otherwise to chain input buffer 9153 if (idx_insert == size) { 9154 if (idx_insert != 0) { 9155 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9156 mEffects[idx_insert-1]->configure(); 9157 } 9158 effect->setOutBuffer(mOutBuffer); 9159 } else { 9160 effect->setOutBuffer(mInBuffer); 9161 } 9162 mEffects.insertAt(effect, idx_insert); 9163 9164 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9165 } 9166 effect->configure(); 9167 return NO_ERROR; 9168} 9169 9170// removeEffect_l() must be called with PlaybackThread::mLock held 9171size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9172{ 9173 Mutex::Autolock _l(mLock); 9174 size_t size = mEffects.size(); 9175 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9176 9177 for (size_t i = 0; i < size; i++) { 9178 if (effect == mEffects[i]) { 9179 // calling stop here will remove pre-processing effect from the audio HAL. 9180 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9181 // the middle of a read from audio HAL 9182 if (mEffects[i]->state() == EffectModule::ACTIVE || 9183 mEffects[i]->state() == EffectModule::STOPPING) { 9184 mEffects[i]->stop(); 9185 } 9186 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9187 delete[] effect->inBuffer(); 9188 } else { 9189 if (i == size - 1 && i != 0) { 9190 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9191 mEffects[i - 1]->configure(); 9192 } 9193 } 9194 mEffects.removeAt(i); 9195 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9196 break; 9197 } 9198 } 9199 9200 return mEffects.size(); 9201} 9202 9203// setDevice_l() must be called with PlaybackThread::mLock held 9204void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9205{ 9206 size_t size = mEffects.size(); 9207 for (size_t i = 0; i < size; i++) { 9208 mEffects[i]->setDevice(device); 9209 } 9210} 9211 9212// setMode_l() must be called with PlaybackThread::mLock held 9213void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9214{ 9215 size_t size = mEffects.size(); 9216 for (size_t i = 0; i < size; i++) { 9217 mEffects[i]->setMode(mode); 9218 } 9219} 9220 9221// setVolume_l() must be called with PlaybackThread::mLock held 9222bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9223{ 9224 uint32_t newLeft = *left; 9225 uint32_t newRight = *right; 9226 bool hasControl = false; 9227 int ctrlIdx = -1; 9228 size_t size = mEffects.size(); 9229 9230 // first update volume controller 9231 for (size_t i = size; i > 0; i--) { 9232 if (mEffects[i - 1]->isProcessEnabled() && 9233 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9234 ctrlIdx = i - 1; 9235 hasControl = true; 9236 break; 9237 } 9238 } 9239 9240 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9241 if (hasControl) { 9242 *left = mNewLeftVolume; 9243 *right = mNewRightVolume; 9244 } 9245 return hasControl; 9246 } 9247 9248 mVolumeCtrlIdx = ctrlIdx; 9249 mLeftVolume = newLeft; 9250 mRightVolume = newRight; 9251 9252 // second get volume update from volume controller 9253 if (ctrlIdx >= 0) { 9254 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9255 mNewLeftVolume = newLeft; 9256 mNewRightVolume = newRight; 9257 } 9258 // then indicate volume to all other effects in chain. 9259 // Pass altered volume to effects before volume controller 9260 // and requested volume to effects after controller 9261 uint32_t lVol = newLeft; 9262 uint32_t rVol = newRight; 9263 9264 for (size_t i = 0; i < size; i++) { 9265 if ((int)i == ctrlIdx) continue; 9266 // this also works for ctrlIdx == -1 when there is no volume controller 9267 if ((int)i > ctrlIdx) { 9268 lVol = *left; 9269 rVol = *right; 9270 } 9271 mEffects[i]->setVolume(&lVol, &rVol, false); 9272 } 9273 *left = newLeft; 9274 *right = newRight; 9275 9276 return hasControl; 9277} 9278 9279status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9280{ 9281 const size_t SIZE = 256; 9282 char buffer[SIZE]; 9283 String8 result; 9284 9285 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9286 result.append(buffer); 9287 9288 bool locked = tryLock(mLock); 9289 // failed to lock - AudioFlinger is probably deadlocked 9290 if (!locked) { 9291 result.append("\tCould not lock mutex:\n"); 9292 } 9293 9294 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9295 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9296 mEffects.size(), 9297 (uint32_t)mInBuffer, 9298 (uint32_t)mOutBuffer, 9299 mActiveTrackCnt); 9300 result.append(buffer); 9301 write(fd, result.string(), result.size()); 9302 9303 for (size_t i = 0; i < mEffects.size(); ++i) { 9304 sp<EffectModule> effect = mEffects[i]; 9305 if (effect != 0) { 9306 effect->dump(fd, args); 9307 } 9308 } 9309 9310 if (locked) { 9311 mLock.unlock(); 9312 } 9313 9314 return NO_ERROR; 9315} 9316 9317// must be called with ThreadBase::mLock held 9318void AudioFlinger::EffectChain::setEffectSuspended_l( 9319 const effect_uuid_t *type, bool suspend) 9320{ 9321 sp<SuspendedEffectDesc> desc; 9322 // use effect type UUID timelow as key as there is no real risk of identical 9323 // timeLow fields among effect type UUIDs. 9324 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9325 if (suspend) { 9326 if (index >= 0) { 9327 desc = mSuspendedEffects.valueAt(index); 9328 } else { 9329 desc = new SuspendedEffectDesc(); 9330 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9331 mSuspendedEffects.add(type->timeLow, desc); 9332 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9333 } 9334 if (desc->mRefCount++ == 0) { 9335 sp<EffectModule> effect = getEffectIfEnabled(type); 9336 if (effect != 0) { 9337 desc->mEffect = effect; 9338 effect->setSuspended(true); 9339 effect->setEnabled(false); 9340 } 9341 } 9342 } else { 9343 if (index < 0) { 9344 return; 9345 } 9346 desc = mSuspendedEffects.valueAt(index); 9347 if (desc->mRefCount <= 0) { 9348 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9349 desc->mRefCount = 1; 9350 } 9351 if (--desc->mRefCount == 0) { 9352 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9353 if (desc->mEffect != 0) { 9354 sp<EffectModule> effect = desc->mEffect.promote(); 9355 if (effect != 0) { 9356 effect->setSuspended(false); 9357 sp<EffectHandle> handle = effect->controlHandle(); 9358 if (handle != 0) { 9359 effect->setEnabled(handle->enabled()); 9360 } 9361 } 9362 desc->mEffect.clear(); 9363 } 9364 mSuspendedEffects.removeItemsAt(index); 9365 } 9366 } 9367} 9368 9369// must be called with ThreadBase::mLock held 9370void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9371{ 9372 sp<SuspendedEffectDesc> desc; 9373 9374 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9375 if (suspend) { 9376 if (index >= 0) { 9377 desc = mSuspendedEffects.valueAt(index); 9378 } else { 9379 desc = new SuspendedEffectDesc(); 9380 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9381 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9382 } 9383 if (desc->mRefCount++ == 0) { 9384 Vector< sp<EffectModule> > effects; 9385 getSuspendEligibleEffects(effects); 9386 for (size_t i = 0; i < effects.size(); i++) { 9387 setEffectSuspended_l(&effects[i]->desc().type, true); 9388 } 9389 } 9390 } else { 9391 if (index < 0) { 9392 return; 9393 } 9394 desc = mSuspendedEffects.valueAt(index); 9395 if (desc->mRefCount <= 0) { 9396 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9397 desc->mRefCount = 1; 9398 } 9399 if (--desc->mRefCount == 0) { 9400 Vector<const effect_uuid_t *> types; 9401 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9402 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9403 continue; 9404 } 9405 types.add(&mSuspendedEffects.valueAt(i)->mType); 9406 } 9407 for (size_t i = 0; i < types.size(); i++) { 9408 setEffectSuspended_l(types[i], false); 9409 } 9410 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9411 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9412 } 9413 } 9414} 9415 9416 9417// The volume effect is used for automated tests only 9418#ifndef OPENSL_ES_H_ 9419static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9420 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9421const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9422#endif //OPENSL_ES_H_ 9423 9424bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9425{ 9426 // auxiliary effects and visualizer are never suspended on output mix 9427 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9428 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9429 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9430 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9431 return false; 9432 } 9433 return true; 9434} 9435 9436void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9437{ 9438 effects.clear(); 9439 for (size_t i = 0; i < mEffects.size(); i++) { 9440 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9441 effects.add(mEffects[i]); 9442 } 9443 } 9444} 9445 9446sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9447 const effect_uuid_t *type) 9448{ 9449 sp<EffectModule> effect = getEffectFromType_l(type); 9450 return effect != 0 && effect->isEnabled() ? effect : 0; 9451} 9452 9453void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9454 bool enabled) 9455{ 9456 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9457 if (enabled) { 9458 if (index < 0) { 9459 // if the effect is not suspend check if all effects are suspended 9460 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9461 if (index < 0) { 9462 return; 9463 } 9464 if (!isEffectEligibleForSuspend(effect->desc())) { 9465 return; 9466 } 9467 setEffectSuspended_l(&effect->desc().type, enabled); 9468 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9469 if (index < 0) { 9470 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9471 return; 9472 } 9473 } 9474 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9475 effect->desc().type.timeLow); 9476 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9477 // if effect is requested to suspended but was not yet enabled, supend it now. 9478 if (desc->mEffect == 0) { 9479 desc->mEffect = effect; 9480 effect->setEnabled(false); 9481 effect->setSuspended(true); 9482 } 9483 } else { 9484 if (index < 0) { 9485 return; 9486 } 9487 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9488 effect->desc().type.timeLow); 9489 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9490 desc->mEffect.clear(); 9491 effect->setSuspended(false); 9492 } 9493} 9494 9495#undef LOG_TAG 9496#define LOG_TAG "AudioFlinger" 9497 9498// ---------------------------------------------------------------------------- 9499 9500status_t AudioFlinger::onTransact( 9501 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9502{ 9503 return BnAudioFlinger::onTransact(code, data, reply, flags); 9504} 9505 9506}; // namespace android 9507