AudioFlinger.cpp revision 510ba8b812d88f62968a2c9b0b638fff6d99ee84
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 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2845 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2846 FastTrack *fastTrack = &state->mFastTracks[j]; 2847 2848 // Determine whether the track is currently in underrun condition, 2849 // and whether it had a recent underrun. 2850 FastTrackUnderruns underruns = mFastMixerDumpState.mTracks[j].mUnderruns; 2851 uint32_t recentFull = (underruns.mBitFields.mFull - 2852 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2853 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2854 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2855 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2856 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2857 uint32_t recentUnderruns = recentPartial + recentEmpty; 2858 track->mObservedUnderruns = underruns; 2859 // don't count underruns that occur while stopping or pausing 2860 // or stopped which can occur when flush() is called while active 2861 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2862 track->mUnderrunCount += recentUnderruns; 2863 } 2864 2865 // This is similar to the state machine for normal tracks, 2866 // with a few modifications for fast tracks. 2867 bool isActive = true; 2868 switch (track->mState) { 2869 case TrackBase::STOPPING_1: 2870 // track stays active in STOPPING_1 state until first underrun 2871 if (recentUnderruns > 0) { 2872 track->mState = TrackBase::STOPPING_2; 2873 } 2874 break; 2875 case TrackBase::PAUSING: 2876 // ramp down is not yet implemented 2877 track->setPaused(); 2878 break; 2879 case TrackBase::RESUMING: 2880 // ramp up is not yet implemented 2881 track->mState = TrackBase::ACTIVE; 2882 break; 2883 case TrackBase::ACTIVE: 2884 if (recentFull > 0 || recentPartial > 0) { 2885 // track has provided at least some frames recently: reset retry count 2886 track->mRetryCount = kMaxTrackRetries; 2887 } 2888 if (recentUnderruns == 0) { 2889 // no recent underruns: stay active 2890 break; 2891 } 2892 // there has recently been an underrun of some kind 2893 if (track->sharedBuffer() == 0) { 2894 // were any of the recent underruns "empty" (no frames available)? 2895 if (recentEmpty == 0) { 2896 // no, then ignore the partial underruns as they are allowed indefinitely 2897 break; 2898 } 2899 // there has recently been an "empty" underrun: decrement the retry counter 2900 if (--(track->mRetryCount) > 0) { 2901 break; 2902 } 2903 // indicate to client process that the track was disabled because of underrun; 2904 // it will then automatically call start() when data is available 2905 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2906 // remove from active list, but state remains ACTIVE [confusing but true] 2907 isActive = false; 2908 break; 2909 } 2910 // fall through 2911 case TrackBase::STOPPING_2: 2912 case TrackBase::PAUSED: 2913 case TrackBase::TERMINATED: 2914 case TrackBase::STOPPED: 2915 case TrackBase::FLUSHED: // flush() while active 2916 // Check for presentation complete if track is inactive 2917 // We have consumed all the buffers of this track. 2918 // This would be incomplete if we auto-paused on underrun 2919 { 2920 size_t audioHALFrames = 2921 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2922 size_t framesWritten = 2923 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2924 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2925 // track stays in active list until presentation is complete 2926 break; 2927 } 2928 } 2929 if (track->isStopping_2()) { 2930 track->mState = TrackBase::STOPPED; 2931 } 2932 if (track->isStopped()) { 2933 // Can't reset directly, as fast mixer is still polling this track 2934 // track->reset(); 2935 // So instead mark this track as needing to be reset after push with ack 2936 resetMask |= 1 << i; 2937 } 2938 isActive = false; 2939 break; 2940 case TrackBase::IDLE: 2941 default: 2942 LOG_FATAL("unexpected track state %d", track->mState); 2943 } 2944 2945 if (isActive) { 2946 // was it previously inactive? 2947 if (!(state->mTrackMask & (1 << j))) { 2948 ExtendedAudioBufferProvider *eabp = track; 2949 VolumeProvider *vp = track; 2950 fastTrack->mBufferProvider = eabp; 2951 fastTrack->mVolumeProvider = vp; 2952 fastTrack->mSampleRate = track->mSampleRate; 2953 fastTrack->mChannelMask = track->mChannelMask; 2954 fastTrack->mGeneration++; 2955 state->mTrackMask |= 1 << j; 2956 didModify = true; 2957 // no acknowledgement required for newly active tracks 2958 } 2959 // cache the combined master volume and stream type volume for fast mixer; this 2960 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2961 track->mCachedVolume = track->isMuted() ? 2962 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2963 ++fastTracks; 2964 } else { 2965 // was it previously active? 2966 if (state->mTrackMask & (1 << j)) { 2967 fastTrack->mBufferProvider = NULL; 2968 fastTrack->mGeneration++; 2969 state->mTrackMask &= ~(1 << j); 2970 didModify = true; 2971 // If any fast tracks were removed, we must wait for acknowledgement 2972 // because we're about to decrement the last sp<> on those tracks. 2973 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2974 } else { 2975 LOG_FATAL("fast track %d should have been active", j); 2976 } 2977 tracksToRemove->add(track); 2978 // Avoids a misleading display in dumpsys 2979 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2980 } 2981 continue; 2982 } 2983 2984 { // local variable scope to avoid goto warning 2985 2986 audio_track_cblk_t* cblk = track->cblk(); 2987 2988 // The first time a track is added we wait 2989 // for all its buffers to be filled before processing it 2990 int name = track->name(); 2991 // make sure that we have enough frames to mix one full buffer. 2992 // enforce this condition only once to enable draining the buffer in case the client 2993 // app does not call stop() and relies on underrun to stop: 2994 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2995 // during last round 2996 uint32_t minFrames = 1; 2997 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2998 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2999 if (t->sampleRate() == (int)mSampleRate) { 3000 minFrames = mNormalFrameCount; 3001 } else { 3002 // +1 for rounding and +1 for additional sample needed for interpolation 3003 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3004 // add frames already consumed but not yet released by the resampler 3005 // because cblk->framesReady() will include these frames 3006 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3007 // the minimum track buffer size is normally twice the number of frames necessary 3008 // to fill one buffer and the resampler should not leave more than one buffer worth 3009 // of unreleased frames after each pass, but just in case... 3010 ALOG_ASSERT(minFrames <= cblk->frameCount); 3011 } 3012 } 3013 if ((track->framesReady() >= minFrames) && track->isReady() && 3014 !track->isPaused() && !track->isTerminated()) 3015 { 3016 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3017 3018 mixedTracks++; 3019 3020 // track->mainBuffer() != mMixBuffer means there is an effect chain 3021 // connected to the track 3022 chain.clear(); 3023 if (track->mainBuffer() != mMixBuffer) { 3024 chain = getEffectChain_l(track->sessionId()); 3025 // Delegate volume control to effect in track effect chain if needed 3026 if (chain != 0) { 3027 tracksWithEffect++; 3028 } else { 3029 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3030 name, track->sessionId()); 3031 } 3032 } 3033 3034 3035 int param = AudioMixer::VOLUME; 3036 if (track->mFillingUpStatus == Track::FS_FILLED) { 3037 // no ramp for the first volume setting 3038 track->mFillingUpStatus = Track::FS_ACTIVE; 3039 if (track->mState == TrackBase::RESUMING) { 3040 track->mState = TrackBase::ACTIVE; 3041 param = AudioMixer::RAMP_VOLUME; 3042 } 3043 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3044 } else if (cblk->server != 0) { 3045 // If the track is stopped before the first frame was mixed, 3046 // do not apply ramp 3047 param = AudioMixer::RAMP_VOLUME; 3048 } 3049 3050 // compute volume for this track 3051 uint32_t vl, vr, va; 3052 if (track->isMuted() || track->isPausing() || 3053 mStreamTypes[track->streamType()].mute) { 3054 vl = vr = va = 0; 3055 if (track->isPausing()) { 3056 track->setPaused(); 3057 } 3058 } else { 3059 3060 // read original volumes with volume control 3061 float typeVolume = mStreamTypes[track->streamType()].volume; 3062 float v = masterVolume * typeVolume; 3063 uint32_t vlr = cblk->getVolumeLR(); 3064 vl = vlr & 0xFFFF; 3065 vr = vlr >> 16; 3066 // track volumes come from shared memory, so can't be trusted and must be clamped 3067 if (vl > MAX_GAIN_INT) { 3068 ALOGV("Track left volume out of range: %04X", vl); 3069 vl = MAX_GAIN_INT; 3070 } 3071 if (vr > MAX_GAIN_INT) { 3072 ALOGV("Track right volume out of range: %04X", vr); 3073 vr = MAX_GAIN_INT; 3074 } 3075 // now apply the master volume and stream type volume 3076 vl = (uint32_t)(v * vl) << 12; 3077 vr = (uint32_t)(v * vr) << 12; 3078 // assuming master volume and stream type volume each go up to 1.0, 3079 // vl and vr are now in 8.24 format 3080 3081 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3082 // send level comes from shared memory and so may be corrupt 3083 if (sendLevel > MAX_GAIN_INT) { 3084 ALOGV("Track send level out of range: %04X", sendLevel); 3085 sendLevel = MAX_GAIN_INT; 3086 } 3087 va = (uint32_t)(v * sendLevel); 3088 } 3089 // Delegate volume control to effect in track effect chain if needed 3090 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3091 // Do not ramp volume if volume is controlled by effect 3092 param = AudioMixer::VOLUME; 3093 track->mHasVolumeController = true; 3094 } else { 3095 // force no volume ramp when volume controller was just disabled or removed 3096 // from effect chain to avoid volume spike 3097 if (track->mHasVolumeController) { 3098 param = AudioMixer::VOLUME; 3099 } 3100 track->mHasVolumeController = false; 3101 } 3102 3103 // Convert volumes from 8.24 to 4.12 format 3104 // This additional clamping is needed in case chain->setVolume_l() overshot 3105 vl = (vl + (1 << 11)) >> 12; 3106 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3107 vr = (vr + (1 << 11)) >> 12; 3108 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3109 3110 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3111 3112 // XXX: these things DON'T need to be done each time 3113 mAudioMixer->setBufferProvider(name, track); 3114 mAudioMixer->enable(name); 3115 3116 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3117 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3118 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3119 mAudioMixer->setParameter( 3120 name, 3121 AudioMixer::TRACK, 3122 AudioMixer::FORMAT, (void *)track->format()); 3123 mAudioMixer->setParameter( 3124 name, 3125 AudioMixer::TRACK, 3126 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3127 mAudioMixer->setParameter( 3128 name, 3129 AudioMixer::RESAMPLE, 3130 AudioMixer::SAMPLE_RATE, 3131 (void *)(cblk->sampleRate)); 3132 mAudioMixer->setParameter( 3133 name, 3134 AudioMixer::TRACK, 3135 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3136 mAudioMixer->setParameter( 3137 name, 3138 AudioMixer::TRACK, 3139 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3140 3141 // reset retry count 3142 track->mRetryCount = kMaxTrackRetries; 3143 3144 // If one track is ready, set the mixer ready if: 3145 // - the mixer was not ready during previous round OR 3146 // - no other track is not ready 3147 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3148 mixerStatus != MIXER_TRACKS_ENABLED) { 3149 mixerStatus = MIXER_TRACKS_READY; 3150 } 3151 } else { 3152 // clear effect chain input buffer if an active track underruns to avoid sending 3153 // previous audio buffer again to effects 3154 chain = getEffectChain_l(track->sessionId()); 3155 if (chain != 0) { 3156 chain->clearInputBuffer(); 3157 } 3158 3159 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3160 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3161 track->isStopped() || track->isPaused()) { 3162 // We have consumed all the buffers of this track. 3163 // Remove it from the list of active tracks. 3164 // TODO: use actual buffer filling status instead of latency when available from 3165 // audio HAL 3166 size_t audioHALFrames = 3167 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3168 size_t framesWritten = 3169 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3170 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3171 if (track->isStopped()) { 3172 track->reset(); 3173 } 3174 tracksToRemove->add(track); 3175 } 3176 } else { 3177 track->mUnderrunCount++; 3178 // No buffers for this track. Give it a few chances to 3179 // fill a buffer, then remove it from active list. 3180 if (--(track->mRetryCount) <= 0) { 3181 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3182 tracksToRemove->add(track); 3183 // indicate to client process that the track was disabled because of underrun; 3184 // it will then automatically call start() when data is available 3185 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3186 // If one track is not ready, mark the mixer also not ready if: 3187 // - the mixer was ready during previous round OR 3188 // - no other track is ready 3189 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3190 mixerStatus != MIXER_TRACKS_READY) { 3191 mixerStatus = MIXER_TRACKS_ENABLED; 3192 } 3193 } 3194 mAudioMixer->disable(name); 3195 } 3196 3197 } // local variable scope to avoid goto warning 3198track_is_ready: ; 3199 3200 } 3201 3202 // Push the new FastMixer state if necessary 3203 if (didModify) { 3204 state->mFastTracksGen++; 3205 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3206 if (kUseFastMixer == FastMixer_Dynamic && 3207 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3208 state->mCommand = FastMixerState::COLD_IDLE; 3209 state->mColdFutexAddr = &mFastMixerFutex; 3210 state->mColdGen++; 3211 mFastMixerFutex = 0; 3212 if (kUseFastMixer == FastMixer_Dynamic) { 3213 mNormalSink = mOutputSink; 3214 } 3215 // If we go into cold idle, need to wait for acknowledgement 3216 // so that fast mixer stops doing I/O. 3217 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3218 } 3219 sq->end(); 3220 } 3221 if (sq != NULL) { 3222 sq->end(didModify); 3223 sq->push(block); 3224 } 3225 3226 // Now perform the deferred reset on fast tracks that have stopped 3227 while (resetMask != 0) { 3228 size_t i = __builtin_ctz(resetMask); 3229 ALOG_ASSERT(i < count); 3230 resetMask &= ~(1 << i); 3231 sp<Track> t = mActiveTracks[i].promote(); 3232 if (t == 0) continue; 3233 Track* track = t.get(); 3234 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3235 track->reset(); 3236 } 3237 3238 // remove all the tracks that need to be... 3239 count = tracksToRemove->size(); 3240 if (CC_UNLIKELY(count)) { 3241 for (size_t i=0 ; i<count ; i++) { 3242 const sp<Track>& track = tracksToRemove->itemAt(i); 3243 mActiveTracks.remove(track); 3244 if (track->mainBuffer() != mMixBuffer) { 3245 chain = getEffectChain_l(track->sessionId()); 3246 if (chain != 0) { 3247 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3248 chain->decActiveTrackCnt(); 3249 } 3250 } 3251 if (track->isTerminated()) { 3252 removeTrack_l(track); 3253 } 3254 } 3255 } 3256 3257 // mix buffer must be cleared if all tracks are connected to an 3258 // effect chain as in this case the mixer will not write to 3259 // mix buffer and track effects will accumulate into it 3260 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3261 // FIXME as a performance optimization, should remember previous zero status 3262 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3263 } 3264 3265 // if any fast tracks, then status is ready 3266 mMixerStatusIgnoringFastTracks = mixerStatus; 3267 if (fastTracks > 0) { 3268 mixerStatus = MIXER_TRACKS_READY; 3269 } 3270 return mixerStatus; 3271} 3272 3273/* 3274The derived values that are cached: 3275 - mixBufferSize from frame count * frame size 3276 - activeSleepTime from activeSleepTimeUs() 3277 - idleSleepTime from idleSleepTimeUs() 3278 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3279 - maxPeriod from frame count and sample rate (MIXER only) 3280 3281The parameters that affect these derived values are: 3282 - frame count 3283 - frame size 3284 - sample rate 3285 - device type: A2DP or not 3286 - device latency 3287 - format: PCM or not 3288 - active sleep time 3289 - idle sleep time 3290*/ 3291 3292void AudioFlinger::PlaybackThread::cacheParameters_l() 3293{ 3294 mixBufferSize = mNormalFrameCount * mFrameSize; 3295 activeSleepTime = activeSleepTimeUs(); 3296 idleSleepTime = idleSleepTimeUs(); 3297} 3298 3299void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3300{ 3301 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3302 this, streamType, mTracks.size()); 3303 Mutex::Autolock _l(mLock); 3304 3305 size_t size = mTracks.size(); 3306 for (size_t i = 0; i < size; i++) { 3307 sp<Track> t = mTracks[i]; 3308 if (t->streamType() == streamType) { 3309 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3310 t->mCblk->cv.signal(); 3311 } 3312 } 3313} 3314 3315// getTrackName_l() must be called with ThreadBase::mLock held 3316int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3317{ 3318 return mAudioMixer->getTrackName(channelMask); 3319} 3320 3321// deleteTrackName_l() must be called with ThreadBase::mLock held 3322void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3323{ 3324 ALOGV("remove track (%d) and delete from mixer", name); 3325 mAudioMixer->deleteTrackName(name); 3326} 3327 3328// checkForNewParameters_l() must be called with ThreadBase::mLock held 3329bool AudioFlinger::MixerThread::checkForNewParameters_l() 3330{ 3331 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3332 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3333 bool reconfig = false; 3334 3335 while (!mNewParameters.isEmpty()) { 3336 3337 if (mFastMixer != NULL) { 3338 FastMixerStateQueue *sq = mFastMixer->sq(); 3339 FastMixerState *state = sq->begin(); 3340 if (!(state->mCommand & FastMixerState::IDLE)) { 3341 previousCommand = state->mCommand; 3342 state->mCommand = FastMixerState::HOT_IDLE; 3343 sq->end(); 3344 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3345 } else { 3346 sq->end(false /*didModify*/); 3347 } 3348 } 3349 3350 status_t status = NO_ERROR; 3351 String8 keyValuePair = mNewParameters[0]; 3352 AudioParameter param = AudioParameter(keyValuePair); 3353 int value; 3354 3355 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3356 reconfig = true; 3357 } 3358 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3359 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3360 status = BAD_VALUE; 3361 } else { 3362 reconfig = true; 3363 } 3364 } 3365 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3366 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3367 status = BAD_VALUE; 3368 } else { 3369 reconfig = true; 3370 } 3371 } 3372 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3373 // do not accept frame count changes if tracks are open as the track buffer 3374 // size depends on frame count and correct behavior would not be guaranteed 3375 // if frame count is changed after track creation 3376 if (!mTracks.isEmpty()) { 3377 status = INVALID_OPERATION; 3378 } else { 3379 reconfig = true; 3380 } 3381 } 3382 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3383#ifdef ADD_BATTERY_DATA 3384 // when changing the audio output device, call addBatteryData to notify 3385 // the change 3386 if ((int)mDevice != value) { 3387 uint32_t params = 0; 3388 // check whether speaker is on 3389 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3390 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3391 } 3392 3393 int deviceWithoutSpeaker 3394 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3395 // check if any other device (except speaker) is on 3396 if (value & deviceWithoutSpeaker ) { 3397 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3398 } 3399 3400 if (params != 0) { 3401 addBatteryData(params); 3402 } 3403 } 3404#endif 3405 3406 // forward device change to effects that have requested to be 3407 // aware of attached audio device. 3408 mDevice = (uint32_t)value; 3409 for (size_t i = 0; i < mEffectChains.size(); i++) { 3410 mEffectChains[i]->setDevice_l(mDevice); 3411 } 3412 } 3413 3414 if (status == NO_ERROR) { 3415 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3416 keyValuePair.string()); 3417 if (!mStandby && status == INVALID_OPERATION) { 3418 mOutput->stream->common.standby(&mOutput->stream->common); 3419 mStandby = true; 3420 mBytesWritten = 0; 3421 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3422 keyValuePair.string()); 3423 } 3424 if (status == NO_ERROR && reconfig) { 3425 delete mAudioMixer; 3426 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3427 mAudioMixer = NULL; 3428 readOutputParameters(); 3429 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3430 for (size_t i = 0; i < mTracks.size() ; i++) { 3431 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3432 if (name < 0) break; 3433 mTracks[i]->mName = name; 3434 // limit track sample rate to 2 x new output sample rate 3435 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3436 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3437 } 3438 } 3439 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3440 } 3441 } 3442 3443 mNewParameters.removeAt(0); 3444 3445 mParamStatus = status; 3446 mParamCond.signal(); 3447 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3448 // already timed out waiting for the status and will never signal the condition. 3449 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3450 } 3451 3452 if (!(previousCommand & FastMixerState::IDLE)) { 3453 ALOG_ASSERT(mFastMixer != NULL); 3454 FastMixerStateQueue *sq = mFastMixer->sq(); 3455 FastMixerState *state = sq->begin(); 3456 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3457 state->mCommand = previousCommand; 3458 sq->end(); 3459 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3460 } 3461 3462 return reconfig; 3463} 3464 3465status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3466{ 3467 const size_t SIZE = 256; 3468 char buffer[SIZE]; 3469 String8 result; 3470 3471 PlaybackThread::dumpInternals(fd, args); 3472 3473 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3474 result.append(buffer); 3475 write(fd, result.string(), result.size()); 3476 3477 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3478 FastMixerDumpState copy = mFastMixerDumpState; 3479 copy.dump(fd); 3480 3481 // Write the tee output to a .wav file 3482 NBAIO_Source *teeSource = mTeeSource.get(); 3483 if (teeSource != NULL) { 3484 char teePath[64]; 3485 struct timeval tv; 3486 gettimeofday(&tv, NULL); 3487 struct tm tm; 3488 localtime_r(&tv.tv_sec, &tm); 3489 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3490 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3491 if (teeFd >= 0) { 3492 char wavHeader[44]; 3493 memcpy(wavHeader, 3494 "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", 3495 sizeof(wavHeader)); 3496 NBAIO_Format format = teeSource->format(); 3497 unsigned channelCount = Format_channelCount(format); 3498 ALOG_ASSERT(channelCount <= FCC_2); 3499 unsigned sampleRate = Format_sampleRate(format); 3500 wavHeader[22] = channelCount; // number of channels 3501 wavHeader[24] = sampleRate; // sample rate 3502 wavHeader[25] = sampleRate >> 8; 3503 wavHeader[32] = channelCount * 2; // block alignment 3504 write(teeFd, wavHeader, sizeof(wavHeader)); 3505 size_t total = 0; 3506 bool firstRead = true; 3507 for (;;) { 3508#define TEE_SINK_READ 1024 3509 short buffer[TEE_SINK_READ * FCC_2]; 3510 size_t count = TEE_SINK_READ; 3511 ssize_t actual = teeSource->read(buffer, count); 3512 bool wasFirstRead = firstRead; 3513 firstRead = false; 3514 if (actual <= 0) { 3515 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3516 continue; 3517 } 3518 break; 3519 } 3520 ALOG_ASSERT(actual <= count); 3521 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3522 total += actual; 3523 } 3524 lseek(teeFd, (off_t) 4, SEEK_SET); 3525 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3526 write(teeFd, &temp, sizeof(temp)); 3527 lseek(teeFd, (off_t) 40, SEEK_SET); 3528 temp = total * channelCount * sizeof(short); 3529 write(teeFd, &temp, sizeof(temp)); 3530 close(teeFd); 3531 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3532 } else { 3533 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3534 } 3535 } 3536 3537 return NO_ERROR; 3538} 3539 3540uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3541{ 3542 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3543} 3544 3545uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3546{ 3547 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3548} 3549 3550void AudioFlinger::MixerThread::cacheParameters_l() 3551{ 3552 PlaybackThread::cacheParameters_l(); 3553 3554 // FIXME: Relaxed timing because of a certain device that can't meet latency 3555 // Should be reduced to 2x after the vendor fixes the driver issue 3556 // increase threshold again due to low power audio mode. The way this warning 3557 // threshold is calculated and its usefulness should be reconsidered anyway. 3558 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3559} 3560 3561// ---------------------------------------------------------------------------- 3562AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3563 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3564 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3565 // mLeftVolFloat, mRightVolFloat 3566 // mLeftVolShort, mRightVolShort 3567{ 3568} 3569 3570AudioFlinger::DirectOutputThread::~DirectOutputThread() 3571{ 3572} 3573 3574AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3575 Vector< sp<Track> > *tracksToRemove 3576) 3577{ 3578 sp<Track> trackToRemove; 3579 3580 mixer_state mixerStatus = MIXER_IDLE; 3581 3582 // find out which tracks need to be processed 3583 if (mActiveTracks.size() != 0) { 3584 sp<Track> t = mActiveTracks[0].promote(); 3585 // The track died recently 3586 if (t == 0) return MIXER_IDLE; 3587 3588 Track* const track = t.get(); 3589 audio_track_cblk_t* cblk = track->cblk(); 3590 3591 // The first time a track is added we wait 3592 // for all its buffers to be filled before processing it 3593 if (cblk->framesReady() && track->isReady() && 3594 !track->isPaused() && !track->isTerminated()) 3595 { 3596 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3597 3598 if (track->mFillingUpStatus == Track::FS_FILLED) { 3599 track->mFillingUpStatus = Track::FS_ACTIVE; 3600 mLeftVolFloat = mRightVolFloat = 0; 3601 mLeftVolShort = mRightVolShort = 0; 3602 if (track->mState == TrackBase::RESUMING) { 3603 track->mState = TrackBase::ACTIVE; 3604 rampVolume = true; 3605 } 3606 } else if (cblk->server != 0) { 3607 // If the track is stopped before the first frame was mixed, 3608 // do not apply ramp 3609 rampVolume = true; 3610 } 3611 // compute volume for this track 3612 float left, right; 3613 if (track->isMuted() || mMasterMute || track->isPausing() || 3614 mStreamTypes[track->streamType()].mute) { 3615 left = right = 0; 3616 if (track->isPausing()) { 3617 track->setPaused(); 3618 } 3619 } else { 3620 float typeVolume = mStreamTypes[track->streamType()].volume; 3621 float v = mMasterVolume * typeVolume; 3622 uint32_t vlr = cblk->getVolumeLR(); 3623 float v_clamped = v * (vlr & 0xFFFF); 3624 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3625 left = v_clamped/MAX_GAIN; 3626 v_clamped = v * (vlr >> 16); 3627 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3628 right = v_clamped/MAX_GAIN; 3629 } 3630 3631 if (left != mLeftVolFloat || right != mRightVolFloat) { 3632 mLeftVolFloat = left; 3633 mRightVolFloat = right; 3634 3635 // If audio HAL implements volume control, 3636 // force software volume to nominal value 3637 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3638 left = 1.0f; 3639 right = 1.0f; 3640 } 3641 3642 // Convert volumes from float to 8.24 3643 uint32_t vl = (uint32_t)(left * (1 << 24)); 3644 uint32_t vr = (uint32_t)(right * (1 << 24)); 3645 3646 // Delegate volume control to effect in track effect chain if needed 3647 // only one effect chain can be present on DirectOutputThread, so if 3648 // there is one, the track is connected to it 3649 if (!mEffectChains.isEmpty()) { 3650 // Do not ramp volume if volume is controlled by effect 3651 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3652 rampVolume = false; 3653 } 3654 } 3655 3656 // Convert volumes from 8.24 to 4.12 format 3657 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3658 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3659 leftVol = (uint16_t)v_clamped; 3660 v_clamped = (vr + (1 << 11)) >> 12; 3661 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3662 rightVol = (uint16_t)v_clamped; 3663 } else { 3664 leftVol = mLeftVolShort; 3665 rightVol = mRightVolShort; 3666 rampVolume = false; 3667 } 3668 3669 // reset retry count 3670 track->mRetryCount = kMaxTrackRetriesDirect; 3671 mActiveTrack = t; 3672 mixerStatus = MIXER_TRACKS_READY; 3673 } else { 3674 // clear effect chain input buffer if an active track underruns to avoid sending 3675 // previous audio buffer again to effects 3676 if (!mEffectChains.isEmpty()) { 3677 mEffectChains[0]->clearInputBuffer(); 3678 } 3679 3680 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3681 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3682 // We have consumed all the buffers of this track. 3683 // Remove it from the list of active tracks. 3684 // TODO: implement behavior for compressed audio 3685 size_t audioHALFrames = 3686 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3687 size_t framesWritten = 3688 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3689 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3690 if (track->isStopped()) { 3691 track->reset(); 3692 } 3693 trackToRemove = track; 3694 } 3695 } else { 3696 // No buffers for this track. Give it a few chances to 3697 // fill a buffer, then remove it from active list. 3698 if (--(track->mRetryCount) <= 0) { 3699 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3700 trackToRemove = track; 3701 } else { 3702 mixerStatus = MIXER_TRACKS_ENABLED; 3703 } 3704 } 3705 } 3706 } 3707 3708 // FIXME merge this with similar code for removing multiple tracks 3709 // remove all the tracks that need to be... 3710 if (CC_UNLIKELY(trackToRemove != 0)) { 3711 tracksToRemove->add(trackToRemove); 3712 mActiveTracks.remove(trackToRemove); 3713 if (!mEffectChains.isEmpty()) { 3714 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3715 trackToRemove->sessionId()); 3716 mEffectChains[0]->decActiveTrackCnt(); 3717 } 3718 if (trackToRemove->isTerminated()) { 3719 removeTrack_l(trackToRemove); 3720 } 3721 } 3722 3723 return mixerStatus; 3724} 3725 3726void AudioFlinger::DirectOutputThread::threadLoop_mix() 3727{ 3728 AudioBufferProvider::Buffer buffer; 3729 size_t frameCount = mFrameCount; 3730 int8_t *curBuf = (int8_t *)mMixBuffer; 3731 // output audio to hardware 3732 while (frameCount) { 3733 buffer.frameCount = frameCount; 3734 mActiveTrack->getNextBuffer(&buffer); 3735 if (CC_UNLIKELY(buffer.raw == NULL)) { 3736 memset(curBuf, 0, frameCount * mFrameSize); 3737 break; 3738 } 3739 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3740 frameCount -= buffer.frameCount; 3741 curBuf += buffer.frameCount * mFrameSize; 3742 mActiveTrack->releaseBuffer(&buffer); 3743 } 3744 sleepTime = 0; 3745 standbyTime = systemTime() + standbyDelay; 3746 mActiveTrack.clear(); 3747 3748 // apply volume 3749 3750 // Do not apply volume on compressed audio 3751 if (!audio_is_linear_pcm(mFormat)) { 3752 return; 3753 } 3754 3755 // convert to signed 16 bit before volume calculation 3756 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3757 size_t count = mFrameCount * mChannelCount; 3758 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3759 int16_t *dst = mMixBuffer + count-1; 3760 while (count--) { 3761 *dst-- = (int16_t)(*src--^0x80) << 8; 3762 } 3763 } 3764 3765 frameCount = mFrameCount; 3766 int16_t *out = mMixBuffer; 3767 if (rampVolume) { 3768 if (mChannelCount == 1) { 3769 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3770 int32_t vlInc = d / (int32_t)frameCount; 3771 int32_t vl = ((int32_t)mLeftVolShort << 16); 3772 do { 3773 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3774 out++; 3775 vl += vlInc; 3776 } while (--frameCount); 3777 3778 } else { 3779 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3780 int32_t vlInc = d / (int32_t)frameCount; 3781 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3782 int32_t vrInc = d / (int32_t)frameCount; 3783 int32_t vl = ((int32_t)mLeftVolShort << 16); 3784 int32_t vr = ((int32_t)mRightVolShort << 16); 3785 do { 3786 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3787 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3788 out += 2; 3789 vl += vlInc; 3790 vr += vrInc; 3791 } while (--frameCount); 3792 } 3793 } else { 3794 if (mChannelCount == 1) { 3795 do { 3796 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3797 out++; 3798 } while (--frameCount); 3799 } else { 3800 do { 3801 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3802 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3803 out += 2; 3804 } while (--frameCount); 3805 } 3806 } 3807 3808 // convert back to unsigned 8 bit after volume calculation 3809 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3810 size_t count = mFrameCount * mChannelCount; 3811 int16_t *src = mMixBuffer; 3812 uint8_t *dst = (uint8_t *)mMixBuffer; 3813 while (count--) { 3814 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3815 } 3816 } 3817 3818 mLeftVolShort = leftVol; 3819 mRightVolShort = rightVol; 3820} 3821 3822void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3823{ 3824 if (sleepTime == 0) { 3825 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3826 sleepTime = activeSleepTime; 3827 } else { 3828 sleepTime = idleSleepTime; 3829 } 3830 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3831 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3832 sleepTime = 0; 3833 } 3834} 3835 3836// getTrackName_l() must be called with ThreadBase::mLock held 3837int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3838{ 3839 return 0; 3840} 3841 3842// deleteTrackName_l() must be called with ThreadBase::mLock held 3843void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3844{ 3845} 3846 3847// checkForNewParameters_l() must be called with ThreadBase::mLock held 3848bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3849{ 3850 bool reconfig = false; 3851 3852 while (!mNewParameters.isEmpty()) { 3853 status_t status = NO_ERROR; 3854 String8 keyValuePair = mNewParameters[0]; 3855 AudioParameter param = AudioParameter(keyValuePair); 3856 int value; 3857 3858 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3859 // do not accept frame count changes if tracks are open as the track buffer 3860 // size depends on frame count and correct behavior would not be garantied 3861 // if frame count is changed after track creation 3862 if (!mTracks.isEmpty()) { 3863 status = INVALID_OPERATION; 3864 } else { 3865 reconfig = true; 3866 } 3867 } 3868 if (status == NO_ERROR) { 3869 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3870 keyValuePair.string()); 3871 if (!mStandby && status == INVALID_OPERATION) { 3872 mOutput->stream->common.standby(&mOutput->stream->common); 3873 mStandby = true; 3874 mBytesWritten = 0; 3875 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3876 keyValuePair.string()); 3877 } 3878 if (status == NO_ERROR && reconfig) { 3879 readOutputParameters(); 3880 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3881 } 3882 } 3883 3884 mNewParameters.removeAt(0); 3885 3886 mParamStatus = status; 3887 mParamCond.signal(); 3888 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3889 // already timed out waiting for the status and will never signal the condition. 3890 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3891 } 3892 return reconfig; 3893} 3894 3895uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3896{ 3897 uint32_t time; 3898 if (audio_is_linear_pcm(mFormat)) { 3899 time = PlaybackThread::activeSleepTimeUs(); 3900 } else { 3901 time = 10000; 3902 } 3903 return time; 3904} 3905 3906uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3907{ 3908 uint32_t time; 3909 if (audio_is_linear_pcm(mFormat)) { 3910 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3911 } else { 3912 time = 10000; 3913 } 3914 return time; 3915} 3916 3917uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3918{ 3919 uint32_t time; 3920 if (audio_is_linear_pcm(mFormat)) { 3921 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3922 } else { 3923 time = 10000; 3924 } 3925 return time; 3926} 3927 3928void AudioFlinger::DirectOutputThread::cacheParameters_l() 3929{ 3930 PlaybackThread::cacheParameters_l(); 3931 3932 // use shorter standby delay as on normal output to release 3933 // hardware resources as soon as possible 3934 standbyDelay = microseconds(activeSleepTime*2); 3935} 3936 3937// ---------------------------------------------------------------------------- 3938 3939AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3940 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3941 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3942 mWaitTimeMs(UINT_MAX) 3943{ 3944 addOutputTrack(mainThread); 3945} 3946 3947AudioFlinger::DuplicatingThread::~DuplicatingThread() 3948{ 3949 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3950 mOutputTracks[i]->destroy(); 3951 } 3952} 3953 3954void AudioFlinger::DuplicatingThread::threadLoop_mix() 3955{ 3956 // mix buffers... 3957 if (outputsReady(outputTracks)) { 3958 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3959 } else { 3960 memset(mMixBuffer, 0, mixBufferSize); 3961 } 3962 sleepTime = 0; 3963 writeFrames = mNormalFrameCount; 3964} 3965 3966void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3967{ 3968 if (sleepTime == 0) { 3969 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3970 sleepTime = activeSleepTime; 3971 } else { 3972 sleepTime = idleSleepTime; 3973 } 3974 } else if (mBytesWritten != 0) { 3975 // flush remaining overflow buffers in output tracks 3976 for (size_t i = 0; i < outputTracks.size(); i++) { 3977 if (outputTracks[i]->isActive()) { 3978 sleepTime = 0; 3979 writeFrames = 0; 3980 memset(mMixBuffer, 0, mixBufferSize); 3981 break; 3982 } 3983 } 3984 } 3985} 3986 3987void AudioFlinger::DuplicatingThread::threadLoop_write() 3988{ 3989 standbyTime = systemTime() + standbyDelay; 3990 for (size_t i = 0; i < outputTracks.size(); i++) { 3991 outputTracks[i]->write(mMixBuffer, writeFrames); 3992 } 3993 mBytesWritten += mixBufferSize; 3994} 3995 3996void AudioFlinger::DuplicatingThread::threadLoop_standby() 3997{ 3998 // DuplicatingThread implements standby by stopping all tracks 3999 for (size_t i = 0; i < outputTracks.size(); i++) { 4000 outputTracks[i]->stop(); 4001 } 4002} 4003 4004void AudioFlinger::DuplicatingThread::saveOutputTracks() 4005{ 4006 outputTracks = mOutputTracks; 4007} 4008 4009void AudioFlinger::DuplicatingThread::clearOutputTracks() 4010{ 4011 outputTracks.clear(); 4012} 4013 4014void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4015{ 4016 Mutex::Autolock _l(mLock); 4017 // FIXME explain this formula 4018 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4019 OutputTrack *outputTrack = new OutputTrack(thread, 4020 this, 4021 mSampleRate, 4022 mFormat, 4023 mChannelMask, 4024 frameCount); 4025 if (outputTrack->cblk() != NULL) { 4026 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4027 mOutputTracks.add(outputTrack); 4028 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4029 updateWaitTime_l(); 4030 } 4031} 4032 4033void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4034{ 4035 Mutex::Autolock _l(mLock); 4036 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4037 if (mOutputTracks[i]->thread() == thread) { 4038 mOutputTracks[i]->destroy(); 4039 mOutputTracks.removeAt(i); 4040 updateWaitTime_l(); 4041 return; 4042 } 4043 } 4044 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4045} 4046 4047// caller must hold mLock 4048void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4049{ 4050 mWaitTimeMs = UINT_MAX; 4051 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4052 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4053 if (strong != 0) { 4054 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4055 if (waitTimeMs < mWaitTimeMs) { 4056 mWaitTimeMs = waitTimeMs; 4057 } 4058 } 4059 } 4060} 4061 4062 4063bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4064{ 4065 for (size_t i = 0; i < outputTracks.size(); i++) { 4066 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4067 if (thread == 0) { 4068 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4069 return false; 4070 } 4071 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4072 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4073 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4074 return false; 4075 } 4076 } 4077 return true; 4078} 4079 4080uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4081{ 4082 return (mWaitTimeMs * 1000) / 2; 4083} 4084 4085void AudioFlinger::DuplicatingThread::cacheParameters_l() 4086{ 4087 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4088 updateWaitTime_l(); 4089 4090 MixerThread::cacheParameters_l(); 4091} 4092 4093// ---------------------------------------------------------------------------- 4094 4095// TrackBase constructor must be called with AudioFlinger::mLock held 4096AudioFlinger::ThreadBase::TrackBase::TrackBase( 4097 ThreadBase *thread, 4098 const sp<Client>& client, 4099 uint32_t sampleRate, 4100 audio_format_t format, 4101 uint32_t channelMask, 4102 int frameCount, 4103 const sp<IMemory>& sharedBuffer, 4104 int sessionId) 4105 : RefBase(), 4106 mThread(thread), 4107 mClient(client), 4108 mCblk(NULL), 4109 // mBuffer 4110 // mBufferEnd 4111 mFrameCount(0), 4112 mState(IDLE), 4113 mSampleRate(sampleRate), 4114 mFormat(format), 4115 mStepServerFailed(false), 4116 mSessionId(sessionId) 4117 // mChannelCount 4118 // mChannelMask 4119{ 4120 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4121 4122 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4123 size_t size = sizeof(audio_track_cblk_t); 4124 uint8_t channelCount = popcount(channelMask); 4125 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4126 if (sharedBuffer == 0) { 4127 size += bufferSize; 4128 } 4129 4130 if (client != NULL) { 4131 mCblkMemory = client->heap()->allocate(size); 4132 if (mCblkMemory != 0) { 4133 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4134 if (mCblk != NULL) { // construct the shared structure in-place. 4135 new(mCblk) audio_track_cblk_t(); 4136 // clear all buffers 4137 mCblk->frameCount = frameCount; 4138 mCblk->sampleRate = sampleRate; 4139// uncomment the following lines to quickly test 32-bit wraparound 4140// mCblk->user = 0xffff0000; 4141// mCblk->server = 0xffff0000; 4142// mCblk->userBase = 0xffff0000; 4143// mCblk->serverBase = 0xffff0000; 4144 mChannelCount = channelCount; 4145 mChannelMask = channelMask; 4146 if (sharedBuffer == 0) { 4147 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4148 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4149 // Force underrun condition to avoid false underrun callback until first data is 4150 // written to buffer (other flags are cleared) 4151 mCblk->flags = CBLK_UNDERRUN_ON; 4152 } else { 4153 mBuffer = sharedBuffer->pointer(); 4154 } 4155 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4156 } 4157 } else { 4158 ALOGE("not enough memory for AudioTrack size=%u", size); 4159 client->heap()->dump("AudioTrack"); 4160 return; 4161 } 4162 } else { 4163 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4164 // construct the shared structure in-place. 4165 new(mCblk) audio_track_cblk_t(); 4166 // clear all buffers 4167 mCblk->frameCount = frameCount; 4168 mCblk->sampleRate = sampleRate; 4169// uncomment the following lines to quickly test 32-bit wraparound 4170// mCblk->user = 0xffff0000; 4171// mCblk->server = 0xffff0000; 4172// mCblk->userBase = 0xffff0000; 4173// mCblk->serverBase = 0xffff0000; 4174 mChannelCount = channelCount; 4175 mChannelMask = channelMask; 4176 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4177 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4178 // Force underrun condition to avoid false underrun callback until first data is 4179 // written to buffer (other flags are cleared) 4180 mCblk->flags = CBLK_UNDERRUN_ON; 4181 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4182 } 4183} 4184 4185AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4186{ 4187 if (mCblk != NULL) { 4188 if (mClient == 0) { 4189 delete mCblk; 4190 } else { 4191 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4192 } 4193 } 4194 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4195 if (mClient != 0) { 4196 // Client destructor must run with AudioFlinger mutex locked 4197 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4198 // If the client's reference count drops to zero, the associated destructor 4199 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4200 // relying on the automatic clear() at end of scope. 4201 mClient.clear(); 4202 } 4203} 4204 4205// AudioBufferProvider interface 4206// getNextBuffer() = 0; 4207// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4208void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4209{ 4210 buffer->raw = NULL; 4211 mFrameCount = buffer->frameCount; 4212 // FIXME See note at getNextBuffer() 4213 (void) step(); // ignore return value of step() 4214 buffer->frameCount = 0; 4215} 4216 4217bool AudioFlinger::ThreadBase::TrackBase::step() { 4218 bool result; 4219 audio_track_cblk_t* cblk = this->cblk(); 4220 4221 result = cblk->stepServer(mFrameCount); 4222 if (!result) { 4223 ALOGV("stepServer failed acquiring cblk mutex"); 4224 mStepServerFailed = true; 4225 } 4226 return result; 4227} 4228 4229void AudioFlinger::ThreadBase::TrackBase::reset() { 4230 audio_track_cblk_t* cblk = this->cblk(); 4231 4232 cblk->user = 0; 4233 cblk->server = 0; 4234 cblk->userBase = 0; 4235 cblk->serverBase = 0; 4236 mStepServerFailed = false; 4237 ALOGV("TrackBase::reset"); 4238} 4239 4240int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4241 return (int)mCblk->sampleRate; 4242} 4243 4244void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4245 audio_track_cblk_t* cblk = this->cblk(); 4246 size_t frameSize = cblk->frameSize; 4247 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4248 int8_t *bufferEnd = bufferStart + frames * frameSize; 4249 4250 // Check validity of returned pointer in case the track control block would have been corrupted. 4251 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4252 "TrackBase::getBuffer buffer out of range:\n" 4253 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4254 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4255 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4256 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4257 4258 return bufferStart; 4259} 4260 4261status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4262{ 4263 mSyncEvents.add(event); 4264 return NO_ERROR; 4265} 4266 4267// ---------------------------------------------------------------------------- 4268 4269// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4270AudioFlinger::PlaybackThread::Track::Track( 4271 PlaybackThread *thread, 4272 const sp<Client>& client, 4273 audio_stream_type_t streamType, 4274 uint32_t sampleRate, 4275 audio_format_t format, 4276 uint32_t channelMask, 4277 int frameCount, 4278 const sp<IMemory>& sharedBuffer, 4279 int sessionId, 4280 IAudioFlinger::track_flags_t flags) 4281 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4282 mMute(false), 4283 mFillingUpStatus(FS_INVALID), 4284 // mRetryCount initialized later when needed 4285 mSharedBuffer(sharedBuffer), 4286 mStreamType(streamType), 4287 mName(-1), // see note below 4288 mMainBuffer(thread->mixBuffer()), 4289 mAuxBuffer(NULL), 4290 mAuxEffectId(0), mHasVolumeController(false), 4291 mPresentationCompleteFrames(0), 4292 mFlags(flags), 4293 mFastIndex(-1), 4294 mUnderrunCount(0), 4295 mCachedVolume(1.0) 4296{ 4297 if (mCblk != NULL) { 4298 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4299 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4300 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4301 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4302 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4303 if (mName < 0) { 4304 ALOGE("no more track names available"); 4305 return; 4306 } 4307 // only allocate a fast track index if we were able to allocate a normal track name 4308 if (flags & IAudioFlinger::TRACK_FAST) { 4309 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4310 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4311 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4312 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4313 // FIXME This is too eager. We allocate a fast track index before the 4314 // fast track becomes active. Since fast tracks are a scarce resource, 4315 // this means we are potentially denying other more important fast tracks from 4316 // being created. It would be better to allocate the index dynamically. 4317 mFastIndex = i; 4318 // Read the initial underruns because this field is never cleared by the fast mixer 4319 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4320 thread->mFastTrackAvailMask &= ~(1 << i); 4321 } 4322 } 4323 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4324} 4325 4326AudioFlinger::PlaybackThread::Track::~Track() 4327{ 4328 ALOGV("PlaybackThread::Track destructor"); 4329 sp<ThreadBase> thread = mThread.promote(); 4330 if (thread != 0) { 4331 Mutex::Autolock _l(thread->mLock); 4332 mState = TERMINATED; 4333 } 4334} 4335 4336void AudioFlinger::PlaybackThread::Track::destroy() 4337{ 4338 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4339 // by removing it from mTracks vector, so there is a risk that this Tracks's 4340 // destructor is called. As the destructor needs to lock mLock, 4341 // we must acquire a strong reference on this Track before locking mLock 4342 // here so that the destructor is called only when exiting this function. 4343 // On the other hand, as long as Track::destroy() is only called by 4344 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4345 // this Track with its member mTrack. 4346 sp<Track> keep(this); 4347 { // scope for mLock 4348 sp<ThreadBase> thread = mThread.promote(); 4349 if (thread != 0) { 4350 if (!isOutputTrack()) { 4351 if (mState == ACTIVE || mState == RESUMING) { 4352 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4353 4354#ifdef ADD_BATTERY_DATA 4355 // to track the speaker usage 4356 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4357#endif 4358 } 4359 AudioSystem::releaseOutput(thread->id()); 4360 } 4361 Mutex::Autolock _l(thread->mLock); 4362 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4363 playbackThread->destroyTrack_l(this); 4364 } 4365 } 4366} 4367 4368/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4369{ 4370 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4371 " Server User Main buf Aux Buf Flags Underruns\n"); 4372} 4373 4374void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4375{ 4376 uint32_t vlr = mCblk->getVolumeLR(); 4377 if (isFastTrack()) { 4378 sprintf(buffer, " F %2d", mFastIndex); 4379 } else { 4380 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4381 } 4382 track_state state = mState; 4383 char stateChar; 4384 switch (state) { 4385 case IDLE: 4386 stateChar = 'I'; 4387 break; 4388 case TERMINATED: 4389 stateChar = 'T'; 4390 break; 4391 case STOPPING_1: 4392 stateChar = 's'; 4393 break; 4394 case STOPPING_2: 4395 stateChar = '5'; 4396 break; 4397 case STOPPED: 4398 stateChar = 'S'; 4399 break; 4400 case RESUMING: 4401 stateChar = 'R'; 4402 break; 4403 case ACTIVE: 4404 stateChar = 'A'; 4405 break; 4406 case PAUSING: 4407 stateChar = 'p'; 4408 break; 4409 case PAUSED: 4410 stateChar = 'P'; 4411 break; 4412 case FLUSHED: 4413 stateChar = 'F'; 4414 break; 4415 default: 4416 stateChar = '?'; 4417 break; 4418 } 4419 char nowInUnderrun; 4420 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4421 case UNDERRUN_FULL: 4422 nowInUnderrun = ' '; 4423 break; 4424 case UNDERRUN_PARTIAL: 4425 nowInUnderrun = '<'; 4426 break; 4427 case UNDERRUN_EMPTY: 4428 nowInUnderrun = '*'; 4429 break; 4430 default: 4431 nowInUnderrun = '?'; 4432 break; 4433 } 4434 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4435 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4436 (mClient == 0) ? getpid_cached : mClient->pid(), 4437 mStreamType, 4438 mFormat, 4439 mChannelMask, 4440 mSessionId, 4441 mFrameCount, 4442 mCblk->frameCount, 4443 stateChar, 4444 mMute, 4445 mFillingUpStatus, 4446 mCblk->sampleRate, 4447 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4448 20.0 * log10((vlr >> 16) / 4096.0), 4449 mCblk->server, 4450 mCblk->user, 4451 (int)mMainBuffer, 4452 (int)mAuxBuffer, 4453 mCblk->flags, 4454 mUnderrunCount, 4455 nowInUnderrun); 4456} 4457 4458// AudioBufferProvider interface 4459status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4460 AudioBufferProvider::Buffer* buffer, int64_t pts) 4461{ 4462 audio_track_cblk_t* cblk = this->cblk(); 4463 uint32_t framesReady; 4464 uint32_t framesReq = buffer->frameCount; 4465 4466 // Check if last stepServer failed, try to step now 4467 if (mStepServerFailed) { 4468 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4469 // Since the fast mixer is higher priority than client callback thread, 4470 // it does not result in priority inversion for client. 4471 // But a non-blocking solution would be preferable to avoid 4472 // fast mixer being unable to tryLock(), and 4473 // to avoid the extra context switches if the client wakes up, 4474 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4475 if (!step()) goto getNextBuffer_exit; 4476 ALOGV("stepServer recovered"); 4477 mStepServerFailed = false; 4478 } 4479 4480 // FIXME Same as above 4481 framesReady = cblk->framesReady(); 4482 4483 if (CC_LIKELY(framesReady)) { 4484 uint32_t s = cblk->server; 4485 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4486 4487 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4488 if (framesReq > framesReady) { 4489 framesReq = framesReady; 4490 } 4491 if (framesReq > bufferEnd - s) { 4492 framesReq = bufferEnd - s; 4493 } 4494 4495 buffer->raw = getBuffer(s, framesReq); 4496 if (buffer->raw == NULL) goto getNextBuffer_exit; 4497 4498 buffer->frameCount = framesReq; 4499 return NO_ERROR; 4500 } 4501 4502getNextBuffer_exit: 4503 buffer->raw = NULL; 4504 buffer->frameCount = 0; 4505 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4506 return NOT_ENOUGH_DATA; 4507} 4508 4509// Note that framesReady() takes a mutex on the control block using tryLock(). 4510// This could result in priority inversion if framesReady() is called by the normal mixer, 4511// as the normal mixer thread runs at lower 4512// priority than the client's callback thread: there is a short window within framesReady() 4513// during which the normal mixer could be preempted, and the client callback would block. 4514// Another problem can occur if framesReady() is called by the fast mixer: 4515// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4516// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4517size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4518 return mCblk->framesReady(); 4519} 4520 4521// Don't call for fast tracks; the framesReady() could result in priority inversion 4522bool AudioFlinger::PlaybackThread::Track::isReady() const { 4523 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4524 4525 if (framesReady() >= mCblk->frameCount || 4526 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4527 mFillingUpStatus = FS_FILLED; 4528 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4529 return true; 4530 } 4531 return false; 4532} 4533 4534status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4535 int triggerSession) 4536{ 4537 status_t status = NO_ERROR; 4538 ALOGV("start(%d), calling pid %d session %d", 4539 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4540 4541 sp<ThreadBase> thread = mThread.promote(); 4542 if (thread != 0) { 4543 Mutex::Autolock _l(thread->mLock); 4544 track_state state = mState; 4545 // here the track could be either new, or restarted 4546 // in both cases "unstop" the track 4547 if (mState == PAUSED) { 4548 mState = TrackBase::RESUMING; 4549 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4550 } else { 4551 mState = TrackBase::ACTIVE; 4552 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4553 } 4554 4555 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4556 thread->mLock.unlock(); 4557 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4558 thread->mLock.lock(); 4559 4560#ifdef ADD_BATTERY_DATA 4561 // to track the speaker usage 4562 if (status == NO_ERROR) { 4563 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4564 } 4565#endif 4566 } 4567 if (status == NO_ERROR) { 4568 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4569 playbackThread->addTrack_l(this); 4570 } else { 4571 mState = state; 4572 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4573 } 4574 } else { 4575 status = BAD_VALUE; 4576 } 4577 return status; 4578} 4579 4580void AudioFlinger::PlaybackThread::Track::stop() 4581{ 4582 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4583 sp<ThreadBase> thread = mThread.promote(); 4584 if (thread != 0) { 4585 Mutex::Autolock _l(thread->mLock); 4586 track_state state = mState; 4587 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4588 // If the track is not active (PAUSED and buffers full), flush buffers 4589 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4590 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4591 reset(); 4592 mState = STOPPED; 4593 } else if (!isFastTrack()) { 4594 mState = STOPPED; 4595 } else { 4596 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4597 // and then to STOPPED and reset() when presentation is complete 4598 mState = STOPPING_1; 4599 } 4600 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4601 } 4602 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4603 thread->mLock.unlock(); 4604 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4605 thread->mLock.lock(); 4606 4607#ifdef ADD_BATTERY_DATA 4608 // to track the speaker usage 4609 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4610#endif 4611 } 4612 } 4613} 4614 4615void AudioFlinger::PlaybackThread::Track::pause() 4616{ 4617 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4618 sp<ThreadBase> thread = mThread.promote(); 4619 if (thread != 0) { 4620 Mutex::Autolock _l(thread->mLock); 4621 if (mState == ACTIVE || mState == RESUMING) { 4622 mState = PAUSING; 4623 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4624 if (!isOutputTrack()) { 4625 thread->mLock.unlock(); 4626 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4627 thread->mLock.lock(); 4628 4629#ifdef ADD_BATTERY_DATA 4630 // to track the speaker usage 4631 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4632#endif 4633 } 4634 } 4635 } 4636} 4637 4638void AudioFlinger::PlaybackThread::Track::flush() 4639{ 4640 ALOGV("flush(%d)", mName); 4641 sp<ThreadBase> thread = mThread.promote(); 4642 if (thread != 0) { 4643 Mutex::Autolock _l(thread->mLock); 4644 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4645 mState != PAUSING) { 4646 return; 4647 } 4648 // No point remaining in PAUSED state after a flush => go to 4649 // FLUSHED state 4650 mState = FLUSHED; 4651 // do not reset the track if it is still in the process of being stopped or paused. 4652 // this will be done by prepareTracks_l() when the track is stopped. 4653 // prepareTracks_l() will see mState == FLUSHED, then 4654 // remove from active track list, reset(), and trigger presentation complete 4655 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4656 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4657 reset(); 4658 } 4659 } 4660} 4661 4662void AudioFlinger::PlaybackThread::Track::reset() 4663{ 4664 // Do not reset twice to avoid discarding data written just after a flush and before 4665 // the audioflinger thread detects the track is stopped. 4666 if (!mResetDone) { 4667 TrackBase::reset(); 4668 // Force underrun condition to avoid false underrun callback until first data is 4669 // written to buffer 4670 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4671 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4672 mFillingUpStatus = FS_FILLING; 4673 mResetDone = true; 4674 if (mState == FLUSHED) { 4675 mState = IDLE; 4676 } 4677 } 4678} 4679 4680void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4681{ 4682 mMute = muted; 4683} 4684 4685status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4686{ 4687 status_t status = DEAD_OBJECT; 4688 sp<ThreadBase> thread = mThread.promote(); 4689 if (thread != 0) { 4690 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4691 status = playbackThread->attachAuxEffect(this, EffectId); 4692 } 4693 return status; 4694} 4695 4696void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4697{ 4698 mAuxEffectId = EffectId; 4699 mAuxBuffer = buffer; 4700} 4701 4702bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4703 size_t audioHalFrames) 4704{ 4705 // a track is considered presented when the total number of frames written to audio HAL 4706 // corresponds to the number of frames written when presentationComplete() is called for the 4707 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4708 if (mPresentationCompleteFrames == 0) { 4709 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4710 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4711 mPresentationCompleteFrames, audioHalFrames); 4712 } 4713 if (framesWritten >= mPresentationCompleteFrames) { 4714 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4715 mSessionId, framesWritten); 4716 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4717 return true; 4718 } 4719 return false; 4720} 4721 4722void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4723{ 4724 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4725 if (mSyncEvents[i]->type() == type) { 4726 mSyncEvents[i]->trigger(); 4727 mSyncEvents.removeAt(i); 4728 i--; 4729 } 4730 } 4731} 4732 4733// implement VolumeBufferProvider interface 4734 4735uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4736{ 4737 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4738 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4739 uint32_t vlr = mCblk->getVolumeLR(); 4740 uint32_t vl = vlr & 0xFFFF; 4741 uint32_t vr = vlr >> 16; 4742 // track volumes come from shared memory, so can't be trusted and must be clamped 4743 if (vl > MAX_GAIN_INT) { 4744 vl = MAX_GAIN_INT; 4745 } 4746 if (vr > MAX_GAIN_INT) { 4747 vr = MAX_GAIN_INT; 4748 } 4749 // now apply the cached master volume and stream type volume; 4750 // this is trusted but lacks any synchronization or barrier so may be stale 4751 float v = mCachedVolume; 4752 vl *= v; 4753 vr *= v; 4754 // re-combine into U4.16 4755 vlr = (vr << 16) | (vl & 0xFFFF); 4756 // FIXME look at mute, pause, and stop flags 4757 return vlr; 4758} 4759 4760status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4761{ 4762 if (mState == TERMINATED || mState == PAUSED || 4763 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4764 (mState == STOPPED)))) { 4765 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4766 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4767 event->cancel(); 4768 return INVALID_OPERATION; 4769 } 4770 TrackBase::setSyncEvent(event); 4771 return NO_ERROR; 4772} 4773 4774// timed audio tracks 4775 4776sp<AudioFlinger::PlaybackThread::TimedTrack> 4777AudioFlinger::PlaybackThread::TimedTrack::create( 4778 PlaybackThread *thread, 4779 const sp<Client>& client, 4780 audio_stream_type_t streamType, 4781 uint32_t sampleRate, 4782 audio_format_t format, 4783 uint32_t channelMask, 4784 int frameCount, 4785 const sp<IMemory>& sharedBuffer, 4786 int sessionId) { 4787 if (!client->reserveTimedTrack()) 4788 return NULL; 4789 4790 return new TimedTrack( 4791 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4792 sharedBuffer, sessionId); 4793} 4794 4795AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4796 PlaybackThread *thread, 4797 const sp<Client>& client, 4798 audio_stream_type_t streamType, 4799 uint32_t sampleRate, 4800 audio_format_t format, 4801 uint32_t channelMask, 4802 int frameCount, 4803 const sp<IMemory>& sharedBuffer, 4804 int sessionId) 4805 : Track(thread, client, streamType, sampleRate, format, channelMask, 4806 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4807 mQueueHeadInFlight(false), 4808 mTrimQueueHeadOnRelease(false), 4809 mFramesPendingInQueue(0), 4810 mTimedSilenceBuffer(NULL), 4811 mTimedSilenceBufferSize(0), 4812 mTimedAudioOutputOnTime(false), 4813 mMediaTimeTransformValid(false) 4814{ 4815 LocalClock lc; 4816 mLocalTimeFreq = lc.getLocalFreq(); 4817 4818 mLocalTimeToSampleTransform.a_zero = 0; 4819 mLocalTimeToSampleTransform.b_zero = 0; 4820 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4821 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4822 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4823 &mLocalTimeToSampleTransform.a_to_b_denom); 4824 4825 mMediaTimeToSampleTransform.a_zero = 0; 4826 mMediaTimeToSampleTransform.b_zero = 0; 4827 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4828 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4829 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4830 &mMediaTimeToSampleTransform.a_to_b_denom); 4831} 4832 4833AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4834 mClient->releaseTimedTrack(); 4835 delete [] mTimedSilenceBuffer; 4836} 4837 4838status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4839 size_t size, sp<IMemory>* buffer) { 4840 4841 Mutex::Autolock _l(mTimedBufferQueueLock); 4842 4843 trimTimedBufferQueue_l(); 4844 4845 // lazily initialize the shared memory heap for timed buffers 4846 if (mTimedMemoryDealer == NULL) { 4847 const int kTimedBufferHeapSize = 512 << 10; 4848 4849 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4850 "AudioFlingerTimed"); 4851 if (mTimedMemoryDealer == NULL) 4852 return NO_MEMORY; 4853 } 4854 4855 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4856 if (newBuffer == NULL) { 4857 newBuffer = mTimedMemoryDealer->allocate(size); 4858 if (newBuffer == NULL) 4859 return NO_MEMORY; 4860 } 4861 4862 *buffer = newBuffer; 4863 return NO_ERROR; 4864} 4865 4866// caller must hold mTimedBufferQueueLock 4867void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4868 int64_t mediaTimeNow; 4869 { 4870 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4871 if (!mMediaTimeTransformValid) 4872 return; 4873 4874 int64_t targetTimeNow; 4875 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4876 ? mCCHelper.getCommonTime(&targetTimeNow) 4877 : mCCHelper.getLocalTime(&targetTimeNow); 4878 4879 if (OK != res) 4880 return; 4881 4882 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4883 &mediaTimeNow)) { 4884 return; 4885 } 4886 } 4887 4888 size_t trimEnd; 4889 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4890 int64_t bufEnd; 4891 4892 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4893 // We have a next buffer. Just use its PTS as the PTS of the frame 4894 // following the last frame in this buffer. If the stream is sparse 4895 // (ie, there are deliberate gaps left in the stream which should be 4896 // filled with silence by the TimedAudioTrack), then this can result 4897 // in one extra buffer being left un-trimmed when it could have 4898 // been. In general, this is not typical, and we would rather 4899 // optimized away the TS calculation below for the more common case 4900 // where PTSes are contiguous. 4901 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4902 } else { 4903 // We have no next buffer. Compute the PTS of the frame following 4904 // the last frame in this buffer by computing the duration of of 4905 // this frame in media time units and adding it to the PTS of the 4906 // buffer. 4907 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4908 / mCblk->frameSize; 4909 4910 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4911 &bufEnd)) { 4912 ALOGE("Failed to convert frame count of %lld to media time" 4913 " duration" " (scale factor %d/%u) in %s", 4914 frameCount, 4915 mMediaTimeToSampleTransform.a_to_b_numer, 4916 mMediaTimeToSampleTransform.a_to_b_denom, 4917 __PRETTY_FUNCTION__); 4918 break; 4919 } 4920 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4921 } 4922 4923 if (bufEnd > mediaTimeNow) 4924 break; 4925 4926 // Is the buffer we want to use in the middle of a mix operation right 4927 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4928 // from the mixer which should be coming back shortly. 4929 if (!trimEnd && mQueueHeadInFlight) { 4930 mTrimQueueHeadOnRelease = true; 4931 } 4932 } 4933 4934 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4935 if (trimStart < trimEnd) { 4936 // Update the bookkeeping for framesReady() 4937 for (size_t i = trimStart; i < trimEnd; ++i) { 4938 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4939 } 4940 4941 // Now actually remove the buffers from the queue. 4942 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4943 } 4944} 4945 4946void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4947 const char* logTag) { 4948 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4949 "%s called (reason \"%s\"), but timed buffer queue has no" 4950 " elements to trim.", __FUNCTION__, logTag); 4951 4952 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4953 mTimedBufferQueue.removeAt(0); 4954} 4955 4956void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4957 const TimedBuffer& buf, 4958 const char* logTag) { 4959 uint32_t bufBytes = buf.buffer()->size(); 4960 uint32_t consumedAlready = buf.position(); 4961 4962 ALOG_ASSERT(consumedAlready <= bufBytes, 4963 "Bad bookkeeping while updating frames pending. Timed buffer is" 4964 " only %u bytes long, but claims to have consumed %u" 4965 " bytes. (update reason: \"%s\")", 4966 bufBytes, consumedAlready, logTag); 4967 4968 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4969 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4970 "Bad bookkeeping while updating frames pending. Should have at" 4971 " least %u queued frames, but we think we have only %u. (update" 4972 " reason: \"%s\")", 4973 bufFrames, mFramesPendingInQueue, logTag); 4974 4975 mFramesPendingInQueue -= bufFrames; 4976} 4977 4978status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4979 const sp<IMemory>& buffer, int64_t pts) { 4980 4981 { 4982 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4983 if (!mMediaTimeTransformValid) 4984 return INVALID_OPERATION; 4985 } 4986 4987 Mutex::Autolock _l(mTimedBufferQueueLock); 4988 4989 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4990 mFramesPendingInQueue += bufFrames; 4991 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4992 4993 return NO_ERROR; 4994} 4995 4996status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4997 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4998 4999 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5000 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5001 target); 5002 5003 if (!(target == TimedAudioTrack::LOCAL_TIME || 5004 target == TimedAudioTrack::COMMON_TIME)) { 5005 return BAD_VALUE; 5006 } 5007 5008 Mutex::Autolock lock(mMediaTimeTransformLock); 5009 mMediaTimeTransform = xform; 5010 mMediaTimeTransformTarget = target; 5011 mMediaTimeTransformValid = true; 5012 5013 return NO_ERROR; 5014} 5015 5016#define min(a, b) ((a) < (b) ? (a) : (b)) 5017 5018// implementation of getNextBuffer for tracks whose buffers have timestamps 5019status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5020 AudioBufferProvider::Buffer* buffer, int64_t pts) 5021{ 5022 if (pts == AudioBufferProvider::kInvalidPTS) { 5023 buffer->raw = 0; 5024 buffer->frameCount = 0; 5025 mTimedAudioOutputOnTime = false; 5026 return INVALID_OPERATION; 5027 } 5028 5029 Mutex::Autolock _l(mTimedBufferQueueLock); 5030 5031 ALOG_ASSERT(!mQueueHeadInFlight, 5032 "getNextBuffer called without releaseBuffer!"); 5033 5034 while (true) { 5035 5036 // if we have no timed buffers, then fail 5037 if (mTimedBufferQueue.isEmpty()) { 5038 buffer->raw = 0; 5039 buffer->frameCount = 0; 5040 return NOT_ENOUGH_DATA; 5041 } 5042 5043 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5044 5045 // calculate the PTS of the head of the timed buffer queue expressed in 5046 // local time 5047 int64_t headLocalPTS; 5048 { 5049 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5050 5051 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5052 5053 if (mMediaTimeTransform.a_to_b_denom == 0) { 5054 // the transform represents a pause, so yield silence 5055 timedYieldSilence_l(buffer->frameCount, buffer); 5056 return NO_ERROR; 5057 } 5058 5059 int64_t transformedPTS; 5060 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5061 &transformedPTS)) { 5062 // the transform failed. this shouldn't happen, but if it does 5063 // then just drop this buffer 5064 ALOGW("timedGetNextBuffer transform failed"); 5065 buffer->raw = 0; 5066 buffer->frameCount = 0; 5067 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5068 return NO_ERROR; 5069 } 5070 5071 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5072 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5073 &headLocalPTS)) { 5074 buffer->raw = 0; 5075 buffer->frameCount = 0; 5076 return INVALID_OPERATION; 5077 } 5078 } else { 5079 headLocalPTS = transformedPTS; 5080 } 5081 } 5082 5083 // adjust the head buffer's PTS to reflect the portion of the head buffer 5084 // that has already been consumed 5085 int64_t effectivePTS = headLocalPTS + 5086 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5087 5088 // Calculate the delta in samples between the head of the input buffer 5089 // queue and the start of the next output buffer that will be written. 5090 // If the transformation fails because of over or underflow, it means 5091 // that the sample's position in the output stream is so far out of 5092 // whack that it should just be dropped. 5093 int64_t sampleDelta; 5094 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5095 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5096 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5097 " mix"); 5098 continue; 5099 } 5100 if (!mLocalTimeToSampleTransform.doForwardTransform( 5101 (effectivePTS - pts) << 32, &sampleDelta)) { 5102 ALOGV("*** too late during sample rate transform: dropped buffer"); 5103 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5104 continue; 5105 } 5106 5107 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5108 " sampleDelta=[%d.%08x]", 5109 head.pts(), head.position(), pts, 5110 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5111 + (sampleDelta >> 32)), 5112 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5113 5114 // if the delta between the ideal placement for the next input sample and 5115 // the current output position is within this threshold, then we will 5116 // concatenate the next input samples to the previous output 5117 const int64_t kSampleContinuityThreshold = 5118 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5119 5120 // if this is the first buffer of audio that we're emitting from this track 5121 // then it should be almost exactly on time. 5122 const int64_t kSampleStartupThreshold = 1LL << 32; 5123 5124 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5125 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5126 // the next input is close enough to being on time, so concatenate it 5127 // with the last output 5128 timedYieldSamples_l(buffer); 5129 5130 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5131 head.position(), buffer->frameCount); 5132 return NO_ERROR; 5133 } 5134 5135 // Looks like our output is not on time. Reset our on timed status. 5136 // Next time we mix samples from our input queue, then should be within 5137 // the StartupThreshold. 5138 mTimedAudioOutputOnTime = false; 5139 if (sampleDelta > 0) { 5140 // the gap between the current output position and the proper start of 5141 // the next input sample is too big, so fill it with silence 5142 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5143 5144 timedYieldSilence_l(framesUntilNextInput, buffer); 5145 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5146 return NO_ERROR; 5147 } else { 5148 // the next input sample is late 5149 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5150 size_t onTimeSamplePosition = 5151 head.position() + lateFrames * mCblk->frameSize; 5152 5153 if (onTimeSamplePosition > head.buffer()->size()) { 5154 // all the remaining samples in the head are too late, so 5155 // drop it and move on 5156 ALOGV("*** too late: dropped buffer"); 5157 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5158 continue; 5159 } else { 5160 // skip over the late samples 5161 head.setPosition(onTimeSamplePosition); 5162 5163 // yield the available samples 5164 timedYieldSamples_l(buffer); 5165 5166 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5167 return NO_ERROR; 5168 } 5169 } 5170 } 5171} 5172 5173// Yield samples from the timed buffer queue head up to the given output 5174// buffer's capacity. 5175// 5176// Caller must hold mTimedBufferQueueLock 5177void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5178 AudioBufferProvider::Buffer* buffer) { 5179 5180 const TimedBuffer& head = mTimedBufferQueue[0]; 5181 5182 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5183 head.position()); 5184 5185 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5186 mCblk->frameSize); 5187 size_t framesRequested = buffer->frameCount; 5188 buffer->frameCount = min(framesLeftInHead, framesRequested); 5189 5190 mQueueHeadInFlight = true; 5191 mTimedAudioOutputOnTime = true; 5192} 5193 5194// Yield samples of silence up to the given output buffer's capacity 5195// 5196// Caller must hold mTimedBufferQueueLock 5197void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5198 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5199 5200 // lazily allocate a buffer filled with silence 5201 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5202 delete [] mTimedSilenceBuffer; 5203 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5204 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5205 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5206 } 5207 5208 buffer->raw = mTimedSilenceBuffer; 5209 size_t framesRequested = buffer->frameCount; 5210 buffer->frameCount = min(numFrames, framesRequested); 5211 5212 mTimedAudioOutputOnTime = false; 5213} 5214 5215// AudioBufferProvider interface 5216void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5217 AudioBufferProvider::Buffer* buffer) { 5218 5219 Mutex::Autolock _l(mTimedBufferQueueLock); 5220 5221 // If the buffer which was just released is part of the buffer at the head 5222 // of the queue, be sure to update the amt of the buffer which has been 5223 // consumed. If the buffer being returned is not part of the head of the 5224 // queue, its either because the buffer is part of the silence buffer, or 5225 // because the head of the timed queue was trimmed after the mixer called 5226 // getNextBuffer but before the mixer called releaseBuffer. 5227 if (buffer->raw == mTimedSilenceBuffer) { 5228 ALOG_ASSERT(!mQueueHeadInFlight, 5229 "Queue head in flight during release of silence buffer!"); 5230 goto done; 5231 } 5232 5233 ALOG_ASSERT(mQueueHeadInFlight, 5234 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5235 " head in flight."); 5236 5237 if (mTimedBufferQueue.size()) { 5238 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5239 5240 void* start = head.buffer()->pointer(); 5241 void* end = reinterpret_cast<void*>( 5242 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5243 + head.buffer()->size()); 5244 5245 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5246 "released buffer not within the head of the timed buffer" 5247 " queue; qHead = [%p, %p], released buffer = %p", 5248 start, end, buffer->raw); 5249 5250 head.setPosition(head.position() + 5251 (buffer->frameCount * mCblk->frameSize)); 5252 mQueueHeadInFlight = false; 5253 5254 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5255 "Bad bookkeeping during releaseBuffer! Should have at" 5256 " least %u queued frames, but we think we have only %u", 5257 buffer->frameCount, mFramesPendingInQueue); 5258 5259 mFramesPendingInQueue -= buffer->frameCount; 5260 5261 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5262 || mTrimQueueHeadOnRelease) { 5263 trimTimedBufferQueueHead_l("releaseBuffer"); 5264 mTrimQueueHeadOnRelease = false; 5265 } 5266 } else { 5267 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5268 " buffers in the timed buffer queue"); 5269 } 5270 5271done: 5272 buffer->raw = 0; 5273 buffer->frameCount = 0; 5274} 5275 5276size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5277 Mutex::Autolock _l(mTimedBufferQueueLock); 5278 return mFramesPendingInQueue; 5279} 5280 5281AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5282 : mPTS(0), mPosition(0) {} 5283 5284AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5285 const sp<IMemory>& buffer, int64_t pts) 5286 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5287 5288// ---------------------------------------------------------------------------- 5289 5290// RecordTrack constructor must be called with AudioFlinger::mLock held 5291AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5292 RecordThread *thread, 5293 const sp<Client>& client, 5294 uint32_t sampleRate, 5295 audio_format_t format, 5296 uint32_t channelMask, 5297 int frameCount, 5298 int sessionId) 5299 : TrackBase(thread, client, sampleRate, format, 5300 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5301 mOverflow(false) 5302{ 5303 if (mCblk != NULL) { 5304 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5305 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5306 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5307 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5308 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5309 } else { 5310 mCblk->frameSize = sizeof(int8_t); 5311 } 5312 } 5313} 5314 5315AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5316{ 5317 sp<ThreadBase> thread = mThread.promote(); 5318 if (thread != 0) { 5319 AudioSystem::releaseInput(thread->id()); 5320 } 5321} 5322 5323// AudioBufferProvider interface 5324status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5325{ 5326 audio_track_cblk_t* cblk = this->cblk(); 5327 uint32_t framesAvail; 5328 uint32_t framesReq = buffer->frameCount; 5329 5330 // Check if last stepServer failed, try to step now 5331 if (mStepServerFailed) { 5332 if (!step()) goto getNextBuffer_exit; 5333 ALOGV("stepServer recovered"); 5334 mStepServerFailed = false; 5335 } 5336 5337 framesAvail = cblk->framesAvailable_l(); 5338 5339 if (CC_LIKELY(framesAvail)) { 5340 uint32_t s = cblk->server; 5341 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5342 5343 if (framesReq > framesAvail) { 5344 framesReq = framesAvail; 5345 } 5346 if (framesReq > bufferEnd - s) { 5347 framesReq = bufferEnd - s; 5348 } 5349 5350 buffer->raw = getBuffer(s, framesReq); 5351 if (buffer->raw == NULL) goto getNextBuffer_exit; 5352 5353 buffer->frameCount = framesReq; 5354 return NO_ERROR; 5355 } 5356 5357getNextBuffer_exit: 5358 buffer->raw = NULL; 5359 buffer->frameCount = 0; 5360 return NOT_ENOUGH_DATA; 5361} 5362 5363status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5364 int triggerSession) 5365{ 5366 sp<ThreadBase> thread = mThread.promote(); 5367 if (thread != 0) { 5368 RecordThread *recordThread = (RecordThread *)thread.get(); 5369 return recordThread->start(this, event, triggerSession); 5370 } else { 5371 return BAD_VALUE; 5372 } 5373} 5374 5375void AudioFlinger::RecordThread::RecordTrack::stop() 5376{ 5377 sp<ThreadBase> thread = mThread.promote(); 5378 if (thread != 0) { 5379 RecordThread *recordThread = (RecordThread *)thread.get(); 5380 recordThread->stop(this); 5381 TrackBase::reset(); 5382 // Force overrun condition to avoid false overrun callback until first data is 5383 // read from buffer 5384 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5385 } 5386} 5387 5388void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5389{ 5390 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5391 (mClient == 0) ? getpid_cached : mClient->pid(), 5392 mFormat, 5393 mChannelMask, 5394 mSessionId, 5395 mFrameCount, 5396 mState, 5397 mCblk->sampleRate, 5398 mCblk->server, 5399 mCblk->user); 5400} 5401 5402 5403// ---------------------------------------------------------------------------- 5404 5405AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5406 PlaybackThread *playbackThread, 5407 DuplicatingThread *sourceThread, 5408 uint32_t sampleRate, 5409 audio_format_t format, 5410 uint32_t channelMask, 5411 int frameCount) 5412 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5413 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5414 mActive(false), mSourceThread(sourceThread) 5415{ 5416 5417 if (mCblk != NULL) { 5418 mCblk->flags |= CBLK_DIRECTION_OUT; 5419 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5420 mOutBuffer.frameCount = 0; 5421 playbackThread->mTracks.add(this); 5422 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5423 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5424 mCblk, mBuffer, mCblk->buffers, 5425 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5426 } else { 5427 ALOGW("Error creating output track on thread %p", playbackThread); 5428 } 5429} 5430 5431AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5432{ 5433 clearBufferQueue(); 5434} 5435 5436status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5437 int triggerSession) 5438{ 5439 status_t status = Track::start(event, triggerSession); 5440 if (status != NO_ERROR) { 5441 return status; 5442 } 5443 5444 mActive = true; 5445 mRetryCount = 127; 5446 return status; 5447} 5448 5449void AudioFlinger::PlaybackThread::OutputTrack::stop() 5450{ 5451 Track::stop(); 5452 clearBufferQueue(); 5453 mOutBuffer.frameCount = 0; 5454 mActive = false; 5455} 5456 5457bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5458{ 5459 Buffer *pInBuffer; 5460 Buffer inBuffer; 5461 uint32_t channelCount = mChannelCount; 5462 bool outputBufferFull = false; 5463 inBuffer.frameCount = frames; 5464 inBuffer.i16 = data; 5465 5466 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5467 5468 if (!mActive && frames != 0) { 5469 start(); 5470 sp<ThreadBase> thread = mThread.promote(); 5471 if (thread != 0) { 5472 MixerThread *mixerThread = (MixerThread *)thread.get(); 5473 if (mCblk->frameCount > frames){ 5474 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5475 uint32_t startFrames = (mCblk->frameCount - frames); 5476 pInBuffer = new Buffer; 5477 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5478 pInBuffer->frameCount = startFrames; 5479 pInBuffer->i16 = pInBuffer->mBuffer; 5480 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5481 mBufferQueue.add(pInBuffer); 5482 } else { 5483 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5484 } 5485 } 5486 } 5487 } 5488 5489 while (waitTimeLeftMs) { 5490 // First write pending buffers, then new data 5491 if (mBufferQueue.size()) { 5492 pInBuffer = mBufferQueue.itemAt(0); 5493 } else { 5494 pInBuffer = &inBuffer; 5495 } 5496 5497 if (pInBuffer->frameCount == 0) { 5498 break; 5499 } 5500 5501 if (mOutBuffer.frameCount == 0) { 5502 mOutBuffer.frameCount = pInBuffer->frameCount; 5503 nsecs_t startTime = systemTime(); 5504 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5505 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5506 outputBufferFull = true; 5507 break; 5508 } 5509 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5510 if (waitTimeLeftMs >= waitTimeMs) { 5511 waitTimeLeftMs -= waitTimeMs; 5512 } else { 5513 waitTimeLeftMs = 0; 5514 } 5515 } 5516 5517 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5518 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5519 mCblk->stepUser(outFrames); 5520 pInBuffer->frameCount -= outFrames; 5521 pInBuffer->i16 += outFrames * channelCount; 5522 mOutBuffer.frameCount -= outFrames; 5523 mOutBuffer.i16 += outFrames * channelCount; 5524 5525 if (pInBuffer->frameCount == 0) { 5526 if (mBufferQueue.size()) { 5527 mBufferQueue.removeAt(0); 5528 delete [] pInBuffer->mBuffer; 5529 delete pInBuffer; 5530 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5531 } else { 5532 break; 5533 } 5534 } 5535 } 5536 5537 // If we could not write all frames, allocate a buffer and queue it for next time. 5538 if (inBuffer.frameCount) { 5539 sp<ThreadBase> thread = mThread.promote(); 5540 if (thread != 0 && !thread->standby()) { 5541 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5542 pInBuffer = new Buffer; 5543 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5544 pInBuffer->frameCount = inBuffer.frameCount; 5545 pInBuffer->i16 = pInBuffer->mBuffer; 5546 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5547 mBufferQueue.add(pInBuffer); 5548 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5549 } else { 5550 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5551 } 5552 } 5553 } 5554 5555 // Calling write() with a 0 length buffer, means that no more data will be written: 5556 // If no more buffers are pending, fill output track buffer to make sure it is started 5557 // by output mixer. 5558 if (frames == 0 && mBufferQueue.size() == 0) { 5559 if (mCblk->user < mCblk->frameCount) { 5560 frames = mCblk->frameCount - mCblk->user; 5561 pInBuffer = new Buffer; 5562 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5563 pInBuffer->frameCount = frames; 5564 pInBuffer->i16 = pInBuffer->mBuffer; 5565 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5566 mBufferQueue.add(pInBuffer); 5567 } else if (mActive) { 5568 stop(); 5569 } 5570 } 5571 5572 return outputBufferFull; 5573} 5574 5575status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5576{ 5577 int active; 5578 status_t result; 5579 audio_track_cblk_t* cblk = mCblk; 5580 uint32_t framesReq = buffer->frameCount; 5581 5582// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5583 buffer->frameCount = 0; 5584 5585 uint32_t framesAvail = cblk->framesAvailable(); 5586 5587 5588 if (framesAvail == 0) { 5589 Mutex::Autolock _l(cblk->lock); 5590 goto start_loop_here; 5591 while (framesAvail == 0) { 5592 active = mActive; 5593 if (CC_UNLIKELY(!active)) { 5594 ALOGV("Not active and NO_MORE_BUFFERS"); 5595 return NO_MORE_BUFFERS; 5596 } 5597 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5598 if (result != NO_ERROR) { 5599 return NO_MORE_BUFFERS; 5600 } 5601 // read the server count again 5602 start_loop_here: 5603 framesAvail = cblk->framesAvailable_l(); 5604 } 5605 } 5606 5607// if (framesAvail < framesReq) { 5608// return NO_MORE_BUFFERS; 5609// } 5610 5611 if (framesReq > framesAvail) { 5612 framesReq = framesAvail; 5613 } 5614 5615 uint32_t u = cblk->user; 5616 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5617 5618 if (framesReq > bufferEnd - u) { 5619 framesReq = bufferEnd - u; 5620 } 5621 5622 buffer->frameCount = framesReq; 5623 buffer->raw = (void *)cblk->buffer(u); 5624 return NO_ERROR; 5625} 5626 5627 5628void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5629{ 5630 size_t size = mBufferQueue.size(); 5631 5632 for (size_t i = 0; i < size; i++) { 5633 Buffer *pBuffer = mBufferQueue.itemAt(i); 5634 delete [] pBuffer->mBuffer; 5635 delete pBuffer; 5636 } 5637 mBufferQueue.clear(); 5638} 5639 5640// ---------------------------------------------------------------------------- 5641 5642AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5643 : RefBase(), 5644 mAudioFlinger(audioFlinger), 5645 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5646 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5647 mPid(pid), 5648 mTimedTrackCount(0) 5649{ 5650 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5651} 5652 5653// Client destructor must be called with AudioFlinger::mLock held 5654AudioFlinger::Client::~Client() 5655{ 5656 mAudioFlinger->removeClient_l(mPid); 5657} 5658 5659sp<MemoryDealer> AudioFlinger::Client::heap() const 5660{ 5661 return mMemoryDealer; 5662} 5663 5664// Reserve one of the limited slots for a timed audio track associated 5665// with this client 5666bool AudioFlinger::Client::reserveTimedTrack() 5667{ 5668 const int kMaxTimedTracksPerClient = 4; 5669 5670 Mutex::Autolock _l(mTimedTrackLock); 5671 5672 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5673 ALOGW("can not create timed track - pid %d has exceeded the limit", 5674 mPid); 5675 return false; 5676 } 5677 5678 mTimedTrackCount++; 5679 return true; 5680} 5681 5682// Release a slot for a timed audio track 5683void AudioFlinger::Client::releaseTimedTrack() 5684{ 5685 Mutex::Autolock _l(mTimedTrackLock); 5686 mTimedTrackCount--; 5687} 5688 5689// ---------------------------------------------------------------------------- 5690 5691AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5692 const sp<IAudioFlingerClient>& client, 5693 pid_t pid) 5694 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5695{ 5696} 5697 5698AudioFlinger::NotificationClient::~NotificationClient() 5699{ 5700} 5701 5702void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5703{ 5704 sp<NotificationClient> keep(this); 5705 mAudioFlinger->removeNotificationClient(mPid); 5706} 5707 5708// ---------------------------------------------------------------------------- 5709 5710AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5711 : BnAudioTrack(), 5712 mTrack(track) 5713{ 5714} 5715 5716AudioFlinger::TrackHandle::~TrackHandle() { 5717 // just stop the track on deletion, associated resources 5718 // will be freed from the main thread once all pending buffers have 5719 // been played. Unless it's not in the active track list, in which 5720 // case we free everything now... 5721 mTrack->destroy(); 5722} 5723 5724sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5725 return mTrack->getCblk(); 5726} 5727 5728status_t AudioFlinger::TrackHandle::start() { 5729 return mTrack->start(); 5730} 5731 5732void AudioFlinger::TrackHandle::stop() { 5733 mTrack->stop(); 5734} 5735 5736void AudioFlinger::TrackHandle::flush() { 5737 mTrack->flush(); 5738} 5739 5740void AudioFlinger::TrackHandle::mute(bool e) { 5741 mTrack->mute(e); 5742} 5743 5744void AudioFlinger::TrackHandle::pause() { 5745 mTrack->pause(); 5746} 5747 5748status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5749{ 5750 return mTrack->attachAuxEffect(EffectId); 5751} 5752 5753status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5754 sp<IMemory>* buffer) { 5755 if (!mTrack->isTimedTrack()) 5756 return INVALID_OPERATION; 5757 5758 PlaybackThread::TimedTrack* tt = 5759 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5760 return tt->allocateTimedBuffer(size, buffer); 5761} 5762 5763status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5764 int64_t pts) { 5765 if (!mTrack->isTimedTrack()) 5766 return INVALID_OPERATION; 5767 5768 PlaybackThread::TimedTrack* tt = 5769 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5770 return tt->queueTimedBuffer(buffer, pts); 5771} 5772 5773status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5774 const LinearTransform& xform, int target) { 5775 5776 if (!mTrack->isTimedTrack()) 5777 return INVALID_OPERATION; 5778 5779 PlaybackThread::TimedTrack* tt = 5780 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5781 return tt->setMediaTimeTransform( 5782 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5783} 5784 5785status_t AudioFlinger::TrackHandle::onTransact( 5786 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5787{ 5788 return BnAudioTrack::onTransact(code, data, reply, flags); 5789} 5790 5791// ---------------------------------------------------------------------------- 5792 5793sp<IAudioRecord> AudioFlinger::openRecord( 5794 pid_t pid, 5795 audio_io_handle_t input, 5796 uint32_t sampleRate, 5797 audio_format_t format, 5798 uint32_t channelMask, 5799 int frameCount, 5800 IAudioFlinger::track_flags_t flags, 5801 int *sessionId, 5802 status_t *status) 5803{ 5804 sp<RecordThread::RecordTrack> recordTrack; 5805 sp<RecordHandle> recordHandle; 5806 sp<Client> client; 5807 status_t lStatus; 5808 RecordThread *thread; 5809 size_t inFrameCount; 5810 int lSessionId; 5811 5812 // check calling permissions 5813 if (!recordingAllowed()) { 5814 lStatus = PERMISSION_DENIED; 5815 goto Exit; 5816 } 5817 5818 // add client to list 5819 { // scope for mLock 5820 Mutex::Autolock _l(mLock); 5821 thread = checkRecordThread_l(input); 5822 if (thread == NULL) { 5823 lStatus = BAD_VALUE; 5824 goto Exit; 5825 } 5826 5827 client = registerPid_l(pid); 5828 5829 // If no audio session id is provided, create one here 5830 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5831 lSessionId = *sessionId; 5832 } else { 5833 lSessionId = nextUniqueId(); 5834 if (sessionId != NULL) { 5835 *sessionId = lSessionId; 5836 } 5837 } 5838 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5839 recordTrack = thread->createRecordTrack_l(client, 5840 sampleRate, 5841 format, 5842 channelMask, 5843 frameCount, 5844 lSessionId, 5845 &lStatus); 5846 } 5847 if (lStatus != NO_ERROR) { 5848 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5849 // destructor is called by the TrackBase destructor with mLock held 5850 client.clear(); 5851 recordTrack.clear(); 5852 goto Exit; 5853 } 5854 5855 // return to handle to client 5856 recordHandle = new RecordHandle(recordTrack); 5857 lStatus = NO_ERROR; 5858 5859Exit: 5860 if (status) { 5861 *status = lStatus; 5862 } 5863 return recordHandle; 5864} 5865 5866// ---------------------------------------------------------------------------- 5867 5868AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5869 : BnAudioRecord(), 5870 mRecordTrack(recordTrack) 5871{ 5872} 5873 5874AudioFlinger::RecordHandle::~RecordHandle() { 5875 stop(); 5876} 5877 5878sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5879 return mRecordTrack->getCblk(); 5880} 5881 5882status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5883 ALOGV("RecordHandle::start()"); 5884 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5885} 5886 5887void AudioFlinger::RecordHandle::stop() { 5888 ALOGV("RecordHandle::stop()"); 5889 mRecordTrack->stop(); 5890} 5891 5892status_t AudioFlinger::RecordHandle::onTransact( 5893 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5894{ 5895 return BnAudioRecord::onTransact(code, data, reply, flags); 5896} 5897 5898// ---------------------------------------------------------------------------- 5899 5900AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5901 AudioStreamIn *input, 5902 uint32_t sampleRate, 5903 uint32_t channels, 5904 audio_io_handle_t id, 5905 uint32_t device) : 5906 ThreadBase(audioFlinger, id, device, RECORD), 5907 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5908 // mRsmpInIndex and mInputBytes set by readInputParameters() 5909 mReqChannelCount(popcount(channels)), 5910 mReqSampleRate(sampleRate) 5911 // mBytesRead is only meaningful while active, and so is cleared in start() 5912 // (but might be better to also clear here for dump?) 5913{ 5914 snprintf(mName, kNameLength, "AudioIn_%X", id); 5915 5916 readInputParameters(); 5917} 5918 5919 5920AudioFlinger::RecordThread::~RecordThread() 5921{ 5922 delete[] mRsmpInBuffer; 5923 delete mResampler; 5924 delete[] mRsmpOutBuffer; 5925} 5926 5927void AudioFlinger::RecordThread::onFirstRef() 5928{ 5929 run(mName, PRIORITY_URGENT_AUDIO); 5930} 5931 5932status_t AudioFlinger::RecordThread::readyToRun() 5933{ 5934 status_t status = initCheck(); 5935 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5936 return status; 5937} 5938 5939bool AudioFlinger::RecordThread::threadLoop() 5940{ 5941 AudioBufferProvider::Buffer buffer; 5942 sp<RecordTrack> activeTrack; 5943 Vector< sp<EffectChain> > effectChains; 5944 5945 nsecs_t lastWarning = 0; 5946 5947 acquireWakeLock(); 5948 5949 // start recording 5950 while (!exitPending()) { 5951 5952 processConfigEvents(); 5953 5954 { // scope for mLock 5955 Mutex::Autolock _l(mLock); 5956 checkForNewParameters_l(); 5957 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5958 if (!mStandby) { 5959 mInput->stream->common.standby(&mInput->stream->common); 5960 mStandby = true; 5961 } 5962 5963 if (exitPending()) break; 5964 5965 releaseWakeLock_l(); 5966 ALOGV("RecordThread: loop stopping"); 5967 // go to sleep 5968 mWaitWorkCV.wait(mLock); 5969 ALOGV("RecordThread: loop starting"); 5970 acquireWakeLock_l(); 5971 continue; 5972 } 5973 if (mActiveTrack != 0) { 5974 if (mActiveTrack->mState == TrackBase::PAUSING) { 5975 if (!mStandby) { 5976 mInput->stream->common.standby(&mInput->stream->common); 5977 mStandby = true; 5978 } 5979 mActiveTrack.clear(); 5980 mStartStopCond.broadcast(); 5981 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5982 if (mReqChannelCount != mActiveTrack->channelCount()) { 5983 mActiveTrack.clear(); 5984 mStartStopCond.broadcast(); 5985 } else if (mBytesRead != 0) { 5986 // record start succeeds only if first read from audio input 5987 // succeeds 5988 if (mBytesRead > 0) { 5989 mActiveTrack->mState = TrackBase::ACTIVE; 5990 } else { 5991 mActiveTrack.clear(); 5992 } 5993 mStartStopCond.broadcast(); 5994 } 5995 mStandby = false; 5996 } 5997 } 5998 lockEffectChains_l(effectChains); 5999 } 6000 6001 if (mActiveTrack != 0) { 6002 if (mActiveTrack->mState != TrackBase::ACTIVE && 6003 mActiveTrack->mState != TrackBase::RESUMING) { 6004 unlockEffectChains(effectChains); 6005 usleep(kRecordThreadSleepUs); 6006 continue; 6007 } 6008 for (size_t i = 0; i < effectChains.size(); i ++) { 6009 effectChains[i]->process_l(); 6010 } 6011 6012 buffer.frameCount = mFrameCount; 6013 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6014 size_t framesOut = buffer.frameCount; 6015 if (mResampler == NULL) { 6016 // no resampling 6017 while (framesOut) { 6018 size_t framesIn = mFrameCount - mRsmpInIndex; 6019 if (framesIn) { 6020 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6021 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6022 if (framesIn > framesOut) 6023 framesIn = framesOut; 6024 mRsmpInIndex += framesIn; 6025 framesOut -= framesIn; 6026 if ((int)mChannelCount == mReqChannelCount || 6027 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6028 memcpy(dst, src, framesIn * mFrameSize); 6029 } else { 6030 int16_t *src16 = (int16_t *)src; 6031 int16_t *dst16 = (int16_t *)dst; 6032 if (mChannelCount == 1) { 6033 while (framesIn--) { 6034 *dst16++ = *src16; 6035 *dst16++ = *src16++; 6036 } 6037 } else { 6038 while (framesIn--) { 6039 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6040 src16 += 2; 6041 } 6042 } 6043 } 6044 } 6045 if (framesOut && mFrameCount == mRsmpInIndex) { 6046 if (framesOut == mFrameCount && 6047 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6048 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6049 framesOut = 0; 6050 } else { 6051 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6052 mRsmpInIndex = 0; 6053 } 6054 if (mBytesRead < 0) { 6055 ALOGE("Error reading audio input"); 6056 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6057 // Force input into standby so that it tries to 6058 // recover at next read attempt 6059 mInput->stream->common.standby(&mInput->stream->common); 6060 usleep(kRecordThreadSleepUs); 6061 } 6062 mRsmpInIndex = mFrameCount; 6063 framesOut = 0; 6064 buffer.frameCount = 0; 6065 } 6066 } 6067 } 6068 } else { 6069 // resampling 6070 6071 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6072 // alter output frame count as if we were expecting stereo samples 6073 if (mChannelCount == 1 && mReqChannelCount == 1) { 6074 framesOut >>= 1; 6075 } 6076 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6077 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6078 // are 32 bit aligned which should be always true. 6079 if (mChannelCount == 2 && mReqChannelCount == 1) { 6080 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6081 // the resampler always outputs stereo samples: do post stereo to mono conversion 6082 int16_t *src = (int16_t *)mRsmpOutBuffer; 6083 int16_t *dst = buffer.i16; 6084 while (framesOut--) { 6085 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6086 src += 2; 6087 } 6088 } else { 6089 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6090 } 6091 6092 } 6093 if (mFramestoDrop == 0) { 6094 mActiveTrack->releaseBuffer(&buffer); 6095 } else { 6096 if (mFramestoDrop > 0) { 6097 mFramestoDrop -= buffer.frameCount; 6098 if (mFramestoDrop <= 0) { 6099 clearSyncStartEvent(); 6100 } 6101 } else { 6102 mFramestoDrop += buffer.frameCount; 6103 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6104 mSyncStartEvent->isCancelled()) { 6105 ALOGW("Synced record %s, session %d, trigger session %d", 6106 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6107 mActiveTrack->sessionId(), 6108 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6109 clearSyncStartEvent(); 6110 } 6111 } 6112 } 6113 mActiveTrack->overflow(); 6114 } 6115 // client isn't retrieving buffers fast enough 6116 else { 6117 if (!mActiveTrack->setOverflow()) { 6118 nsecs_t now = systemTime(); 6119 if ((now - lastWarning) > kWarningThrottleNs) { 6120 ALOGW("RecordThread: buffer overflow"); 6121 lastWarning = now; 6122 } 6123 } 6124 // Release the processor for a while before asking for a new buffer. 6125 // This will give the application more chance to read from the buffer and 6126 // clear the overflow. 6127 usleep(kRecordThreadSleepUs); 6128 } 6129 } 6130 // enable changes in effect chain 6131 unlockEffectChains(effectChains); 6132 effectChains.clear(); 6133 } 6134 6135 if (!mStandby) { 6136 mInput->stream->common.standby(&mInput->stream->common); 6137 } 6138 mActiveTrack.clear(); 6139 6140 mStartStopCond.broadcast(); 6141 6142 releaseWakeLock(); 6143 6144 ALOGV("RecordThread %p exiting", this); 6145 return false; 6146} 6147 6148 6149sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6150 const sp<AudioFlinger::Client>& client, 6151 uint32_t sampleRate, 6152 audio_format_t format, 6153 int channelMask, 6154 int frameCount, 6155 int sessionId, 6156 status_t *status) 6157{ 6158 sp<RecordTrack> track; 6159 status_t lStatus; 6160 6161 lStatus = initCheck(); 6162 if (lStatus != NO_ERROR) { 6163 ALOGE("Audio driver not initialized."); 6164 goto Exit; 6165 } 6166 6167 { // scope for mLock 6168 Mutex::Autolock _l(mLock); 6169 6170 track = new RecordTrack(this, client, sampleRate, 6171 format, channelMask, frameCount, sessionId); 6172 6173 if (track->getCblk() == 0) { 6174 lStatus = NO_MEMORY; 6175 goto Exit; 6176 } 6177 6178 mTrack = track.get(); 6179 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6180 bool suspend = audio_is_bluetooth_sco_device( 6181 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6182 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6183 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6184 } 6185 lStatus = NO_ERROR; 6186 6187Exit: 6188 if (status) { 6189 *status = lStatus; 6190 } 6191 return track; 6192} 6193 6194status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6195 AudioSystem::sync_event_t event, 6196 int triggerSession) 6197{ 6198 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6199 sp<ThreadBase> strongMe = this; 6200 status_t status = NO_ERROR; 6201 6202 if (event == AudioSystem::SYNC_EVENT_NONE) { 6203 clearSyncStartEvent(); 6204 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6205 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6206 triggerSession, 6207 recordTrack->sessionId(), 6208 syncStartEventCallback, 6209 this); 6210 // Sync event can be cancelled by the trigger session if the track is not in a 6211 // compatible state in which case we start record immediately 6212 if (mSyncStartEvent->isCancelled()) { 6213 clearSyncStartEvent(); 6214 } else { 6215 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6216 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6217 } 6218 } 6219 6220 { 6221 AutoMutex lock(mLock); 6222 if (mActiveTrack != 0) { 6223 if (recordTrack != mActiveTrack.get()) { 6224 status = -EBUSY; 6225 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6226 mActiveTrack->mState = TrackBase::ACTIVE; 6227 } 6228 return status; 6229 } 6230 6231 recordTrack->mState = TrackBase::IDLE; 6232 mActiveTrack = recordTrack; 6233 mLock.unlock(); 6234 status_t status = AudioSystem::startInput(mId); 6235 mLock.lock(); 6236 if (status != NO_ERROR) { 6237 mActiveTrack.clear(); 6238 clearSyncStartEvent(); 6239 return status; 6240 } 6241 mRsmpInIndex = mFrameCount; 6242 mBytesRead = 0; 6243 if (mResampler != NULL) { 6244 mResampler->reset(); 6245 } 6246 mActiveTrack->mState = TrackBase::RESUMING; 6247 // signal thread to start 6248 ALOGV("Signal record thread"); 6249 mWaitWorkCV.signal(); 6250 // do not wait for mStartStopCond if exiting 6251 if (exitPending()) { 6252 mActiveTrack.clear(); 6253 status = INVALID_OPERATION; 6254 goto startError; 6255 } 6256 mStartStopCond.wait(mLock); 6257 if (mActiveTrack == 0) { 6258 ALOGV("Record failed to start"); 6259 status = BAD_VALUE; 6260 goto startError; 6261 } 6262 ALOGV("Record started OK"); 6263 return status; 6264 } 6265startError: 6266 AudioSystem::stopInput(mId); 6267 clearSyncStartEvent(); 6268 return status; 6269} 6270 6271void AudioFlinger::RecordThread::clearSyncStartEvent() 6272{ 6273 if (mSyncStartEvent != 0) { 6274 mSyncStartEvent->cancel(); 6275 } 6276 mSyncStartEvent.clear(); 6277 mFramestoDrop = 0; 6278} 6279 6280void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6281{ 6282 sp<SyncEvent> strongEvent = event.promote(); 6283 6284 if (strongEvent != 0) { 6285 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6286 me->handleSyncStartEvent(strongEvent); 6287 } 6288} 6289 6290void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6291{ 6292 if (event == mSyncStartEvent) { 6293 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6294 // from audio HAL 6295 mFramestoDrop = mFrameCount * 2; 6296 } 6297} 6298 6299void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6300 ALOGV("RecordThread::stop"); 6301 sp<ThreadBase> strongMe = this; 6302 { 6303 AutoMutex lock(mLock); 6304 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6305 mActiveTrack->mState = TrackBase::PAUSING; 6306 // do not wait for mStartStopCond if exiting 6307 if (exitPending()) { 6308 return; 6309 } 6310 mStartStopCond.wait(mLock); 6311 // if we have been restarted, recordTrack == mActiveTrack.get() here 6312 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6313 mLock.unlock(); 6314 AudioSystem::stopInput(mId); 6315 mLock.lock(); 6316 ALOGV("Record stopped OK"); 6317 } 6318 } 6319 } 6320} 6321 6322bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6323{ 6324 return false; 6325} 6326 6327status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6328{ 6329 if (!isValidSyncEvent(event)) { 6330 return BAD_VALUE; 6331 } 6332 6333 Mutex::Autolock _l(mLock); 6334 6335 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6336 mTrack->setSyncEvent(event); 6337 return NO_ERROR; 6338 } 6339 return NAME_NOT_FOUND; 6340} 6341 6342status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6343{ 6344 const size_t SIZE = 256; 6345 char buffer[SIZE]; 6346 String8 result; 6347 6348 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6349 result.append(buffer); 6350 6351 if (mActiveTrack != 0) { 6352 result.append("Active Track:\n"); 6353 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6354 mActiveTrack->dump(buffer, SIZE); 6355 result.append(buffer); 6356 6357 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6358 result.append(buffer); 6359 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6360 result.append(buffer); 6361 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6362 result.append(buffer); 6363 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6364 result.append(buffer); 6365 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6366 result.append(buffer); 6367 6368 6369 } else { 6370 result.append("No record client\n"); 6371 } 6372 write(fd, result.string(), result.size()); 6373 6374 dumpBase(fd, args); 6375 dumpEffectChains(fd, args); 6376 6377 return NO_ERROR; 6378} 6379 6380// AudioBufferProvider interface 6381status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6382{ 6383 size_t framesReq = buffer->frameCount; 6384 size_t framesReady = mFrameCount - mRsmpInIndex; 6385 int channelCount; 6386 6387 if (framesReady == 0) { 6388 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6389 if (mBytesRead < 0) { 6390 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6391 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6392 // Force input into standby so that it tries to 6393 // recover at next read attempt 6394 mInput->stream->common.standby(&mInput->stream->common); 6395 usleep(kRecordThreadSleepUs); 6396 } 6397 buffer->raw = NULL; 6398 buffer->frameCount = 0; 6399 return NOT_ENOUGH_DATA; 6400 } 6401 mRsmpInIndex = 0; 6402 framesReady = mFrameCount; 6403 } 6404 6405 if (framesReq > framesReady) { 6406 framesReq = framesReady; 6407 } 6408 6409 if (mChannelCount == 1 && mReqChannelCount == 2) { 6410 channelCount = 1; 6411 } else { 6412 channelCount = 2; 6413 } 6414 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6415 buffer->frameCount = framesReq; 6416 return NO_ERROR; 6417} 6418 6419// AudioBufferProvider interface 6420void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6421{ 6422 mRsmpInIndex += buffer->frameCount; 6423 buffer->frameCount = 0; 6424} 6425 6426bool AudioFlinger::RecordThread::checkForNewParameters_l() 6427{ 6428 bool reconfig = false; 6429 6430 while (!mNewParameters.isEmpty()) { 6431 status_t status = NO_ERROR; 6432 String8 keyValuePair = mNewParameters[0]; 6433 AudioParameter param = AudioParameter(keyValuePair); 6434 int value; 6435 audio_format_t reqFormat = mFormat; 6436 int reqSamplingRate = mReqSampleRate; 6437 int reqChannelCount = mReqChannelCount; 6438 6439 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6440 reqSamplingRate = value; 6441 reconfig = true; 6442 } 6443 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6444 reqFormat = (audio_format_t) value; 6445 reconfig = true; 6446 } 6447 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6448 reqChannelCount = popcount(value); 6449 reconfig = true; 6450 } 6451 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6452 // do not accept frame count changes if tracks are open as the track buffer 6453 // size depends on frame count and correct behavior would not be guaranteed 6454 // if frame count is changed after track creation 6455 if (mActiveTrack != 0) { 6456 status = INVALID_OPERATION; 6457 } else { 6458 reconfig = true; 6459 } 6460 } 6461 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6462 // forward device change to effects that have requested to be 6463 // aware of attached audio device. 6464 for (size_t i = 0; i < mEffectChains.size(); i++) { 6465 mEffectChains[i]->setDevice_l(value); 6466 } 6467 // store input device and output device but do not forward output device to audio HAL. 6468 // Note that status is ignored by the caller for output device 6469 // (see AudioFlinger::setParameters() 6470 if (value & AUDIO_DEVICE_OUT_ALL) { 6471 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6472 status = BAD_VALUE; 6473 } else { 6474 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6475 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6476 if (mTrack != NULL) { 6477 bool suspend = audio_is_bluetooth_sco_device( 6478 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6479 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6480 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6481 } 6482 } 6483 mDevice |= (uint32_t)value; 6484 } 6485 if (status == NO_ERROR) { 6486 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6487 if (status == INVALID_OPERATION) { 6488 mInput->stream->common.standby(&mInput->stream->common); 6489 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6490 keyValuePair.string()); 6491 } 6492 if (reconfig) { 6493 if (status == BAD_VALUE && 6494 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6495 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6496 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6497 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6498 (reqChannelCount <= FCC_2)) { 6499 status = NO_ERROR; 6500 } 6501 if (status == NO_ERROR) { 6502 readInputParameters(); 6503 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6504 } 6505 } 6506 } 6507 6508 mNewParameters.removeAt(0); 6509 6510 mParamStatus = status; 6511 mParamCond.signal(); 6512 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6513 // already timed out waiting for the status and will never signal the condition. 6514 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6515 } 6516 return reconfig; 6517} 6518 6519String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6520{ 6521 char *s; 6522 String8 out_s8 = String8(); 6523 6524 Mutex::Autolock _l(mLock); 6525 if (initCheck() != NO_ERROR) { 6526 return out_s8; 6527 } 6528 6529 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6530 out_s8 = String8(s); 6531 free(s); 6532 return out_s8; 6533} 6534 6535void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6536 AudioSystem::OutputDescriptor desc; 6537 void *param2 = NULL; 6538 6539 switch (event) { 6540 case AudioSystem::INPUT_OPENED: 6541 case AudioSystem::INPUT_CONFIG_CHANGED: 6542 desc.channels = mChannelMask; 6543 desc.samplingRate = mSampleRate; 6544 desc.format = mFormat; 6545 desc.frameCount = mFrameCount; 6546 desc.latency = 0; 6547 param2 = &desc; 6548 break; 6549 6550 case AudioSystem::INPUT_CLOSED: 6551 default: 6552 break; 6553 } 6554 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6555} 6556 6557void AudioFlinger::RecordThread::readInputParameters() 6558{ 6559 delete mRsmpInBuffer; 6560 // mRsmpInBuffer is always assigned a new[] below 6561 delete mRsmpOutBuffer; 6562 mRsmpOutBuffer = NULL; 6563 delete mResampler; 6564 mResampler = NULL; 6565 6566 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6567 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6568 mChannelCount = (uint16_t)popcount(mChannelMask); 6569 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6570 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6571 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6572 mFrameCount = mInputBytes / mFrameSize; 6573 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6574 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6575 6576 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6577 { 6578 int channelCount; 6579 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6580 // stereo to mono post process as the resampler always outputs stereo. 6581 if (mChannelCount == 1 && mReqChannelCount == 2) { 6582 channelCount = 1; 6583 } else { 6584 channelCount = 2; 6585 } 6586 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6587 mResampler->setSampleRate(mSampleRate); 6588 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6589 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6590 6591 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6592 if (mChannelCount == 1 && mReqChannelCount == 1) { 6593 mFrameCount >>= 1; 6594 } 6595 6596 } 6597 mRsmpInIndex = mFrameCount; 6598} 6599 6600unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6601{ 6602 Mutex::Autolock _l(mLock); 6603 if (initCheck() != NO_ERROR) { 6604 return 0; 6605 } 6606 6607 return mInput->stream->get_input_frames_lost(mInput->stream); 6608} 6609 6610uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6611{ 6612 Mutex::Autolock _l(mLock); 6613 uint32_t result = 0; 6614 if (getEffectChain_l(sessionId) != 0) { 6615 result = EFFECT_SESSION; 6616 } 6617 6618 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6619 result |= TRACK_SESSION; 6620 } 6621 6622 return result; 6623} 6624 6625AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6626{ 6627 Mutex::Autolock _l(mLock); 6628 return mTrack; 6629} 6630 6631AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6632{ 6633 Mutex::Autolock _l(mLock); 6634 return mInput; 6635} 6636 6637AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6638{ 6639 Mutex::Autolock _l(mLock); 6640 AudioStreamIn *input = mInput; 6641 mInput = NULL; 6642 return input; 6643} 6644 6645// this method must always be called either with ThreadBase mLock held or inside the thread loop 6646audio_stream_t* AudioFlinger::RecordThread::stream() const 6647{ 6648 if (mInput == NULL) { 6649 return NULL; 6650 } 6651 return &mInput->stream->common; 6652} 6653 6654 6655// ---------------------------------------------------------------------------- 6656 6657audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6658{ 6659 if (!settingsAllowed()) { 6660 return 0; 6661 } 6662 Mutex::Autolock _l(mLock); 6663 return loadHwModule_l(name); 6664} 6665 6666// loadHwModule_l() must be called with AudioFlinger::mLock held 6667audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6668{ 6669 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6670 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6671 ALOGW("loadHwModule() module %s already loaded", name); 6672 return mAudioHwDevs.keyAt(i); 6673 } 6674 } 6675 6676 audio_hw_device_t *dev; 6677 6678 int rc = load_audio_interface(name, &dev); 6679 if (rc) { 6680 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6681 return 0; 6682 } 6683 6684 mHardwareStatus = AUDIO_HW_INIT; 6685 rc = dev->init_check(dev); 6686 mHardwareStatus = AUDIO_HW_IDLE; 6687 if (rc) { 6688 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6689 return 0; 6690 } 6691 6692 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6693 (NULL != dev->set_master_volume)) { 6694 AutoMutex lock(mHardwareLock); 6695 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6696 dev->set_master_volume(dev, mMasterVolume); 6697 mHardwareStatus = AUDIO_HW_IDLE; 6698 } 6699 6700 audio_module_handle_t handle = nextUniqueId(); 6701 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6702 6703 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6704 name, dev->common.module->name, dev->common.module->id, handle); 6705 6706 return handle; 6707 6708} 6709 6710audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6711 audio_devices_t *pDevices, 6712 uint32_t *pSamplingRate, 6713 audio_format_t *pFormat, 6714 audio_channel_mask_t *pChannelMask, 6715 uint32_t *pLatencyMs, 6716 audio_output_flags_t flags) 6717{ 6718 status_t status; 6719 PlaybackThread *thread = NULL; 6720 struct audio_config config = { 6721 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6722 channel_mask: pChannelMask ? *pChannelMask : 0, 6723 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6724 }; 6725 audio_stream_out_t *outStream = NULL; 6726 audio_hw_device_t *outHwDev; 6727 6728 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6729 module, 6730 (pDevices != NULL) ? (int)*pDevices : 0, 6731 config.sample_rate, 6732 config.format, 6733 config.channel_mask, 6734 flags); 6735 6736 if (pDevices == NULL || *pDevices == 0) { 6737 return 0; 6738 } 6739 6740 Mutex::Autolock _l(mLock); 6741 6742 outHwDev = findSuitableHwDev_l(module, *pDevices); 6743 if (outHwDev == NULL) 6744 return 0; 6745 6746 audio_io_handle_t id = nextUniqueId(); 6747 6748 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6749 6750 status = outHwDev->open_output_stream(outHwDev, 6751 id, 6752 *pDevices, 6753 (audio_output_flags_t)flags, 6754 &config, 6755 &outStream); 6756 6757 mHardwareStatus = AUDIO_HW_IDLE; 6758 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6759 outStream, 6760 config.sample_rate, 6761 config.format, 6762 config.channel_mask, 6763 status); 6764 6765 if (status == NO_ERROR && outStream != NULL) { 6766 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6767 6768 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6769 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6770 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6771 thread = new DirectOutputThread(this, output, id, *pDevices); 6772 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6773 } else { 6774 thread = new MixerThread(this, output, id, *pDevices); 6775 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6776 } 6777 mPlaybackThreads.add(id, thread); 6778 6779 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6780 if (pFormat != NULL) *pFormat = config.format; 6781 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6782 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6783 6784 // notify client processes of the new output creation 6785 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6786 6787 // the first primary output opened designates the primary hw device 6788 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6789 ALOGI("Using module %d has the primary audio interface", module); 6790 mPrimaryHardwareDev = outHwDev; 6791 6792 AutoMutex lock(mHardwareLock); 6793 mHardwareStatus = AUDIO_HW_SET_MODE; 6794 outHwDev->set_mode(outHwDev, mMode); 6795 6796 // Determine the level of master volume support the primary audio HAL has, 6797 // and set the initial master volume at the same time. 6798 float initialVolume = 1.0; 6799 mMasterVolumeSupportLvl = MVS_NONE; 6800 6801 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6802 if ((NULL != outHwDev->get_master_volume) && 6803 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6804 mMasterVolumeSupportLvl = MVS_FULL; 6805 } else { 6806 mMasterVolumeSupportLvl = MVS_SETONLY; 6807 initialVolume = 1.0; 6808 } 6809 6810 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6811 if ((NULL == outHwDev->set_master_volume) || 6812 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6813 mMasterVolumeSupportLvl = MVS_NONE; 6814 } 6815 // now that we have a primary device, initialize master volume on other devices 6816 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6817 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6818 6819 if ((dev != mPrimaryHardwareDev) && 6820 (NULL != dev->set_master_volume)) { 6821 dev->set_master_volume(dev, initialVolume); 6822 } 6823 } 6824 mHardwareStatus = AUDIO_HW_IDLE; 6825 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6826 ? initialVolume 6827 : 1.0; 6828 mMasterVolume = initialVolume; 6829 } 6830 return id; 6831 } 6832 6833 return 0; 6834} 6835 6836audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6837 audio_io_handle_t output2) 6838{ 6839 Mutex::Autolock _l(mLock); 6840 MixerThread *thread1 = checkMixerThread_l(output1); 6841 MixerThread *thread2 = checkMixerThread_l(output2); 6842 6843 if (thread1 == NULL || thread2 == NULL) { 6844 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6845 return 0; 6846 } 6847 6848 audio_io_handle_t id = nextUniqueId(); 6849 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6850 thread->addOutputTrack(thread2); 6851 mPlaybackThreads.add(id, thread); 6852 // notify client processes of the new output creation 6853 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6854 return id; 6855} 6856 6857status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6858{ 6859 // keep strong reference on the playback thread so that 6860 // it is not destroyed while exit() is executed 6861 sp<PlaybackThread> thread; 6862 { 6863 Mutex::Autolock _l(mLock); 6864 thread = checkPlaybackThread_l(output); 6865 if (thread == NULL) { 6866 return BAD_VALUE; 6867 } 6868 6869 ALOGV("closeOutput() %d", output); 6870 6871 if (thread->type() == ThreadBase::MIXER) { 6872 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6873 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6874 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6875 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6876 } 6877 } 6878 } 6879 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6880 mPlaybackThreads.removeItem(output); 6881 } 6882 thread->exit(); 6883 // The thread entity (active unit of execution) is no longer running here, 6884 // but the ThreadBase container still exists. 6885 6886 if (thread->type() != ThreadBase::DUPLICATING) { 6887 AudioStreamOut *out = thread->clearOutput(); 6888 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6889 // from now on thread->mOutput is NULL 6890 out->hwDev->close_output_stream(out->hwDev, out->stream); 6891 delete out; 6892 } 6893 return NO_ERROR; 6894} 6895 6896status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6897{ 6898 Mutex::Autolock _l(mLock); 6899 PlaybackThread *thread = checkPlaybackThread_l(output); 6900 6901 if (thread == NULL) { 6902 return BAD_VALUE; 6903 } 6904 6905 ALOGV("suspendOutput() %d", output); 6906 thread->suspend(); 6907 6908 return NO_ERROR; 6909} 6910 6911status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6912{ 6913 Mutex::Autolock _l(mLock); 6914 PlaybackThread *thread = checkPlaybackThread_l(output); 6915 6916 if (thread == NULL) { 6917 return BAD_VALUE; 6918 } 6919 6920 ALOGV("restoreOutput() %d", output); 6921 6922 thread->restore(); 6923 6924 return NO_ERROR; 6925} 6926 6927audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6928 audio_devices_t *pDevices, 6929 uint32_t *pSamplingRate, 6930 audio_format_t *pFormat, 6931 uint32_t *pChannelMask) 6932{ 6933 status_t status; 6934 RecordThread *thread = NULL; 6935 struct audio_config config = { 6936 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6937 channel_mask: pChannelMask ? *pChannelMask : 0, 6938 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6939 }; 6940 uint32_t reqSamplingRate = config.sample_rate; 6941 audio_format_t reqFormat = config.format; 6942 audio_channel_mask_t reqChannels = config.channel_mask; 6943 audio_stream_in_t *inStream = NULL; 6944 audio_hw_device_t *inHwDev; 6945 6946 if (pDevices == NULL || *pDevices == 0) { 6947 return 0; 6948 } 6949 6950 Mutex::Autolock _l(mLock); 6951 6952 inHwDev = findSuitableHwDev_l(module, *pDevices); 6953 if (inHwDev == NULL) 6954 return 0; 6955 6956 audio_io_handle_t id = nextUniqueId(); 6957 6958 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6959 &inStream); 6960 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6961 inStream, 6962 config.sample_rate, 6963 config.format, 6964 config.channel_mask, 6965 status); 6966 6967 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6968 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6969 // or stereo to mono conversions on 16 bit PCM inputs. 6970 if (status == BAD_VALUE && 6971 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6972 (config.sample_rate <= 2 * reqSamplingRate) && 6973 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6974 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6975 inStream = NULL; 6976 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6977 } 6978 6979 if (status == NO_ERROR && inStream != NULL) { 6980 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6981 6982 // Start record thread 6983 // RecorThread require both input and output device indication to forward to audio 6984 // pre processing modules 6985 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6986 thread = new RecordThread(this, 6987 input, 6988 reqSamplingRate, 6989 reqChannels, 6990 id, 6991 device); 6992 mRecordThreads.add(id, thread); 6993 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6994 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6995 if (pFormat != NULL) *pFormat = config.format; 6996 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6997 6998 input->stream->common.standby(&input->stream->common); 6999 7000 // notify client processes of the new input creation 7001 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7002 return id; 7003 } 7004 7005 return 0; 7006} 7007 7008status_t AudioFlinger::closeInput(audio_io_handle_t input) 7009{ 7010 // keep strong reference on the record thread so that 7011 // it is not destroyed while exit() is executed 7012 sp<RecordThread> thread; 7013 { 7014 Mutex::Autolock _l(mLock); 7015 thread = checkRecordThread_l(input); 7016 if (thread == NULL) { 7017 return BAD_VALUE; 7018 } 7019 7020 ALOGV("closeInput() %d", input); 7021 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7022 mRecordThreads.removeItem(input); 7023 } 7024 thread->exit(); 7025 // The thread entity (active unit of execution) is no longer running here, 7026 // but the ThreadBase container still exists. 7027 7028 AudioStreamIn *in = thread->clearInput(); 7029 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7030 // from now on thread->mInput is NULL 7031 in->hwDev->close_input_stream(in->hwDev, in->stream); 7032 delete in; 7033 7034 return NO_ERROR; 7035} 7036 7037status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7038{ 7039 Mutex::Autolock _l(mLock); 7040 MixerThread *dstThread = checkMixerThread_l(output); 7041 if (dstThread == NULL) { 7042 ALOGW("setStreamOutput() bad output id %d", output); 7043 return BAD_VALUE; 7044 } 7045 7046 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7047 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7048 7049 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7050 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7051 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7052 MixerThread *srcThread = (MixerThread *)thread; 7053 srcThread->invalidateTracks(stream); 7054 } 7055 } 7056 7057 return NO_ERROR; 7058} 7059 7060 7061int AudioFlinger::newAudioSessionId() 7062{ 7063 return nextUniqueId(); 7064} 7065 7066void AudioFlinger::acquireAudioSessionId(int audioSession) 7067{ 7068 Mutex::Autolock _l(mLock); 7069 pid_t caller = IPCThreadState::self()->getCallingPid(); 7070 ALOGV("acquiring %d from %d", audioSession, caller); 7071 size_t num = mAudioSessionRefs.size(); 7072 for (size_t i = 0; i< num; i++) { 7073 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7074 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7075 ref->mCnt++; 7076 ALOGV(" incremented refcount to %d", ref->mCnt); 7077 return; 7078 } 7079 } 7080 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7081 ALOGV(" added new entry for %d", audioSession); 7082} 7083 7084void AudioFlinger::releaseAudioSessionId(int audioSession) 7085{ 7086 Mutex::Autolock _l(mLock); 7087 pid_t caller = IPCThreadState::self()->getCallingPid(); 7088 ALOGV("releasing %d from %d", audioSession, caller); 7089 size_t num = mAudioSessionRefs.size(); 7090 for (size_t i = 0; i< num; i++) { 7091 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7092 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7093 ref->mCnt--; 7094 ALOGV(" decremented refcount to %d", ref->mCnt); 7095 if (ref->mCnt == 0) { 7096 mAudioSessionRefs.removeAt(i); 7097 delete ref; 7098 purgeStaleEffects_l(); 7099 } 7100 return; 7101 } 7102 } 7103 ALOGW("session id %d not found for pid %d", audioSession, caller); 7104} 7105 7106void AudioFlinger::purgeStaleEffects_l() { 7107 7108 ALOGV("purging stale effects"); 7109 7110 Vector< sp<EffectChain> > chains; 7111 7112 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7113 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7114 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7115 sp<EffectChain> ec = t->mEffectChains[j]; 7116 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7117 chains.push(ec); 7118 } 7119 } 7120 } 7121 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7122 sp<RecordThread> t = mRecordThreads.valueAt(i); 7123 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7124 sp<EffectChain> ec = t->mEffectChains[j]; 7125 chains.push(ec); 7126 } 7127 } 7128 7129 for (size_t i = 0; i < chains.size(); i++) { 7130 sp<EffectChain> ec = chains[i]; 7131 int sessionid = ec->sessionId(); 7132 sp<ThreadBase> t = ec->mThread.promote(); 7133 if (t == 0) { 7134 continue; 7135 } 7136 size_t numsessionrefs = mAudioSessionRefs.size(); 7137 bool found = false; 7138 for (size_t k = 0; k < numsessionrefs; k++) { 7139 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7140 if (ref->mSessionid == sessionid) { 7141 ALOGV(" session %d still exists for %d with %d refs", 7142 sessionid, ref->mPid, ref->mCnt); 7143 found = true; 7144 break; 7145 } 7146 } 7147 if (!found) { 7148 // remove all effects from the chain 7149 while (ec->mEffects.size()) { 7150 sp<EffectModule> effect = ec->mEffects[0]; 7151 effect->unPin(); 7152 Mutex::Autolock _l (t->mLock); 7153 t->removeEffect_l(effect); 7154 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7155 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7156 if (handle != 0) { 7157 handle->mEffect.clear(); 7158 if (handle->mHasControl && handle->mEnabled) { 7159 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7160 } 7161 } 7162 } 7163 AudioSystem::unregisterEffect(effect->id()); 7164 } 7165 } 7166 } 7167 return; 7168} 7169 7170// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7171AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7172{ 7173 return mPlaybackThreads.valueFor(output).get(); 7174} 7175 7176// checkMixerThread_l() must be called with AudioFlinger::mLock held 7177AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7178{ 7179 PlaybackThread *thread = checkPlaybackThread_l(output); 7180 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7181} 7182 7183// checkRecordThread_l() must be called with AudioFlinger::mLock held 7184AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7185{ 7186 return mRecordThreads.valueFor(input).get(); 7187} 7188 7189uint32_t AudioFlinger::nextUniqueId() 7190{ 7191 return android_atomic_inc(&mNextUniqueId); 7192} 7193 7194AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7195{ 7196 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7197 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7198 AudioStreamOut *output = thread->getOutput(); 7199 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7200 return thread; 7201 } 7202 } 7203 return NULL; 7204} 7205 7206uint32_t AudioFlinger::primaryOutputDevice_l() const 7207{ 7208 PlaybackThread *thread = primaryPlaybackThread_l(); 7209 7210 if (thread == NULL) { 7211 return 0; 7212 } 7213 7214 return thread->device(); 7215} 7216 7217sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7218 int triggerSession, 7219 int listenerSession, 7220 sync_event_callback_t callBack, 7221 void *cookie) 7222{ 7223 Mutex::Autolock _l(mLock); 7224 7225 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7226 status_t playStatus = NAME_NOT_FOUND; 7227 status_t recStatus = NAME_NOT_FOUND; 7228 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7229 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7230 if (playStatus == NO_ERROR) { 7231 return event; 7232 } 7233 } 7234 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7235 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7236 if (recStatus == NO_ERROR) { 7237 return event; 7238 } 7239 } 7240 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7241 mPendingSyncEvents.add(event); 7242 } else { 7243 ALOGV("createSyncEvent() invalid event %d", event->type()); 7244 event.clear(); 7245 } 7246 return event; 7247} 7248 7249// ---------------------------------------------------------------------------- 7250// Effect management 7251// ---------------------------------------------------------------------------- 7252 7253 7254status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7255{ 7256 Mutex::Autolock _l(mLock); 7257 return EffectQueryNumberEffects(numEffects); 7258} 7259 7260status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7261{ 7262 Mutex::Autolock _l(mLock); 7263 return EffectQueryEffect(index, descriptor); 7264} 7265 7266status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7267 effect_descriptor_t *descriptor) const 7268{ 7269 Mutex::Autolock _l(mLock); 7270 return EffectGetDescriptor(pUuid, descriptor); 7271} 7272 7273 7274sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7275 effect_descriptor_t *pDesc, 7276 const sp<IEffectClient>& effectClient, 7277 int32_t priority, 7278 audio_io_handle_t io, 7279 int sessionId, 7280 status_t *status, 7281 int *id, 7282 int *enabled) 7283{ 7284 status_t lStatus = NO_ERROR; 7285 sp<EffectHandle> handle; 7286 effect_descriptor_t desc; 7287 7288 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7289 pid, effectClient.get(), priority, sessionId, io); 7290 7291 if (pDesc == NULL) { 7292 lStatus = BAD_VALUE; 7293 goto Exit; 7294 } 7295 7296 // check audio settings permission for global effects 7297 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7298 lStatus = PERMISSION_DENIED; 7299 goto Exit; 7300 } 7301 7302 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7303 // that can only be created by audio policy manager (running in same process) 7304 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7305 lStatus = PERMISSION_DENIED; 7306 goto Exit; 7307 } 7308 7309 if (io == 0) { 7310 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7311 // output must be specified by AudioPolicyManager when using session 7312 // AUDIO_SESSION_OUTPUT_STAGE 7313 lStatus = BAD_VALUE; 7314 goto Exit; 7315 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7316 // if the output returned by getOutputForEffect() is removed before we lock the 7317 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7318 // and we will exit safely 7319 io = AudioSystem::getOutputForEffect(&desc); 7320 } 7321 } 7322 7323 { 7324 Mutex::Autolock _l(mLock); 7325 7326 7327 if (!EffectIsNullUuid(&pDesc->uuid)) { 7328 // if uuid is specified, request effect descriptor 7329 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7330 if (lStatus < 0) { 7331 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7332 goto Exit; 7333 } 7334 } else { 7335 // if uuid is not specified, look for an available implementation 7336 // of the required type in effect factory 7337 if (EffectIsNullUuid(&pDesc->type)) { 7338 ALOGW("createEffect() no effect type"); 7339 lStatus = BAD_VALUE; 7340 goto Exit; 7341 } 7342 uint32_t numEffects = 0; 7343 effect_descriptor_t d; 7344 d.flags = 0; // prevent compiler warning 7345 bool found = false; 7346 7347 lStatus = EffectQueryNumberEffects(&numEffects); 7348 if (lStatus < 0) { 7349 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7350 goto Exit; 7351 } 7352 for (uint32_t i = 0; i < numEffects; i++) { 7353 lStatus = EffectQueryEffect(i, &desc); 7354 if (lStatus < 0) { 7355 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7356 continue; 7357 } 7358 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7359 // If matching type found save effect descriptor. If the session is 7360 // 0 and the effect is not auxiliary, continue enumeration in case 7361 // an auxiliary version of this effect type is available 7362 found = true; 7363 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7364 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7365 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7366 break; 7367 } 7368 } 7369 } 7370 if (!found) { 7371 lStatus = BAD_VALUE; 7372 ALOGW("createEffect() effect not found"); 7373 goto Exit; 7374 } 7375 // For same effect type, chose auxiliary version over insert version if 7376 // connect to output mix (Compliance to OpenSL ES) 7377 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7378 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7379 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7380 } 7381 } 7382 7383 // Do not allow auxiliary effects on a session different from 0 (output mix) 7384 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7385 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7386 lStatus = INVALID_OPERATION; 7387 goto Exit; 7388 } 7389 7390 // check recording permission for visualizer 7391 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7392 !recordingAllowed()) { 7393 lStatus = PERMISSION_DENIED; 7394 goto Exit; 7395 } 7396 7397 // return effect descriptor 7398 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7399 7400 // If output is not specified try to find a matching audio session ID in one of the 7401 // output threads. 7402 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7403 // because of code checking output when entering the function. 7404 // Note: io is never 0 when creating an effect on an input 7405 if (io == 0) { 7406 // look for the thread where the specified audio session is present 7407 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7408 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7409 io = mPlaybackThreads.keyAt(i); 7410 break; 7411 } 7412 } 7413 if (io == 0) { 7414 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7415 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7416 io = mRecordThreads.keyAt(i); 7417 break; 7418 } 7419 } 7420 } 7421 // If no output thread contains the requested session ID, default to 7422 // first output. The effect chain will be moved to the correct output 7423 // thread when a track with the same session ID is created 7424 if (io == 0 && mPlaybackThreads.size()) { 7425 io = mPlaybackThreads.keyAt(0); 7426 } 7427 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7428 } 7429 ThreadBase *thread = checkRecordThread_l(io); 7430 if (thread == NULL) { 7431 thread = checkPlaybackThread_l(io); 7432 if (thread == NULL) { 7433 ALOGE("createEffect() unknown output thread"); 7434 lStatus = BAD_VALUE; 7435 goto Exit; 7436 } 7437 } 7438 7439 sp<Client> client = registerPid_l(pid); 7440 7441 // create effect on selected output thread 7442 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7443 &desc, enabled, &lStatus); 7444 if (handle != 0 && id != NULL) { 7445 *id = handle->id(); 7446 } 7447 } 7448 7449Exit: 7450 if (status != NULL) { 7451 *status = lStatus; 7452 } 7453 return handle; 7454} 7455 7456status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7457 audio_io_handle_t dstOutput) 7458{ 7459 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7460 sessionId, srcOutput, dstOutput); 7461 Mutex::Autolock _l(mLock); 7462 if (srcOutput == dstOutput) { 7463 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7464 return NO_ERROR; 7465 } 7466 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7467 if (srcThread == NULL) { 7468 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7469 return BAD_VALUE; 7470 } 7471 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7472 if (dstThread == NULL) { 7473 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7474 return BAD_VALUE; 7475 } 7476 7477 Mutex::Autolock _dl(dstThread->mLock); 7478 Mutex::Autolock _sl(srcThread->mLock); 7479 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7480 7481 return NO_ERROR; 7482} 7483 7484// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7485status_t AudioFlinger::moveEffectChain_l(int sessionId, 7486 AudioFlinger::PlaybackThread *srcThread, 7487 AudioFlinger::PlaybackThread *dstThread, 7488 bool reRegister) 7489{ 7490 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7491 sessionId, srcThread, dstThread); 7492 7493 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7494 if (chain == 0) { 7495 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7496 sessionId, srcThread); 7497 return INVALID_OPERATION; 7498 } 7499 7500 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7501 // so that a new chain is created with correct parameters when first effect is added. This is 7502 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7503 // removed. 7504 srcThread->removeEffectChain_l(chain); 7505 7506 // transfer all effects one by one so that new effect chain is created on new thread with 7507 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7508 audio_io_handle_t dstOutput = dstThread->id(); 7509 sp<EffectChain> dstChain; 7510 uint32_t strategy = 0; // prevent compiler warning 7511 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7512 while (effect != 0) { 7513 srcThread->removeEffect_l(effect); 7514 dstThread->addEffect_l(effect); 7515 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7516 if (effect->state() == EffectModule::ACTIVE || 7517 effect->state() == EffectModule::STOPPING) { 7518 effect->start(); 7519 } 7520 // if the move request is not received from audio policy manager, the effect must be 7521 // re-registered with the new strategy and output 7522 if (dstChain == 0) { 7523 dstChain = effect->chain().promote(); 7524 if (dstChain == 0) { 7525 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7526 srcThread->addEffect_l(effect); 7527 return NO_INIT; 7528 } 7529 strategy = dstChain->strategy(); 7530 } 7531 if (reRegister) { 7532 AudioSystem::unregisterEffect(effect->id()); 7533 AudioSystem::registerEffect(&effect->desc(), 7534 dstOutput, 7535 strategy, 7536 sessionId, 7537 effect->id()); 7538 } 7539 effect = chain->getEffectFromId_l(0); 7540 } 7541 7542 return NO_ERROR; 7543} 7544 7545 7546// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7547sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7548 const sp<AudioFlinger::Client>& client, 7549 const sp<IEffectClient>& effectClient, 7550 int32_t priority, 7551 int sessionId, 7552 effect_descriptor_t *desc, 7553 int *enabled, 7554 status_t *status 7555 ) 7556{ 7557 sp<EffectModule> effect; 7558 sp<EffectHandle> handle; 7559 status_t lStatus; 7560 sp<EffectChain> chain; 7561 bool chainCreated = false; 7562 bool effectCreated = false; 7563 bool effectRegistered = false; 7564 7565 lStatus = initCheck(); 7566 if (lStatus != NO_ERROR) { 7567 ALOGW("createEffect_l() Audio driver not initialized."); 7568 goto Exit; 7569 } 7570 7571 // Do not allow effects with session ID 0 on direct output or duplicating threads 7572 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7573 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7574 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7575 desc->name, sessionId); 7576 lStatus = BAD_VALUE; 7577 goto Exit; 7578 } 7579 // Only Pre processor effects are allowed on input threads and only on input threads 7580 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7581 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7582 desc->name, desc->flags, mType); 7583 lStatus = BAD_VALUE; 7584 goto Exit; 7585 } 7586 7587 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7588 7589 { // scope for mLock 7590 Mutex::Autolock _l(mLock); 7591 7592 // check for existing effect chain with the requested audio session 7593 chain = getEffectChain_l(sessionId); 7594 if (chain == 0) { 7595 // create a new chain for this session 7596 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7597 chain = new EffectChain(this, sessionId); 7598 addEffectChain_l(chain); 7599 chain->setStrategy(getStrategyForSession_l(sessionId)); 7600 chainCreated = true; 7601 } else { 7602 effect = chain->getEffectFromDesc_l(desc); 7603 } 7604 7605 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7606 7607 if (effect == 0) { 7608 int id = mAudioFlinger->nextUniqueId(); 7609 // Check CPU and memory usage 7610 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7611 if (lStatus != NO_ERROR) { 7612 goto Exit; 7613 } 7614 effectRegistered = true; 7615 // create a new effect module if none present in the chain 7616 effect = new EffectModule(this, chain, desc, id, sessionId); 7617 lStatus = effect->status(); 7618 if (lStatus != NO_ERROR) { 7619 goto Exit; 7620 } 7621 lStatus = chain->addEffect_l(effect); 7622 if (lStatus != NO_ERROR) { 7623 goto Exit; 7624 } 7625 effectCreated = true; 7626 7627 effect->setDevice(mDevice); 7628 effect->setMode(mAudioFlinger->getMode()); 7629 } 7630 // create effect handle and connect it to effect module 7631 handle = new EffectHandle(effect, client, effectClient, priority); 7632 lStatus = effect->addHandle(handle); 7633 if (enabled != NULL) { 7634 *enabled = (int)effect->isEnabled(); 7635 } 7636 } 7637 7638Exit: 7639 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7640 Mutex::Autolock _l(mLock); 7641 if (effectCreated) { 7642 chain->removeEffect_l(effect); 7643 } 7644 if (effectRegistered) { 7645 AudioSystem::unregisterEffect(effect->id()); 7646 } 7647 if (chainCreated) { 7648 removeEffectChain_l(chain); 7649 } 7650 handle.clear(); 7651 } 7652 7653 if (status != NULL) { 7654 *status = lStatus; 7655 } 7656 return handle; 7657} 7658 7659sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7660{ 7661 sp<EffectChain> chain = getEffectChain_l(sessionId); 7662 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7663} 7664 7665// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7666// PlaybackThread::mLock held 7667status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7668{ 7669 // check for existing effect chain with the requested audio session 7670 int sessionId = effect->sessionId(); 7671 sp<EffectChain> chain = getEffectChain_l(sessionId); 7672 bool chainCreated = false; 7673 7674 if (chain == 0) { 7675 // create a new chain for this session 7676 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7677 chain = new EffectChain(this, sessionId); 7678 addEffectChain_l(chain); 7679 chain->setStrategy(getStrategyForSession_l(sessionId)); 7680 chainCreated = true; 7681 } 7682 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7683 7684 if (chain->getEffectFromId_l(effect->id()) != 0) { 7685 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7686 this, effect->desc().name, chain.get()); 7687 return BAD_VALUE; 7688 } 7689 7690 status_t status = chain->addEffect_l(effect); 7691 if (status != NO_ERROR) { 7692 if (chainCreated) { 7693 removeEffectChain_l(chain); 7694 } 7695 return status; 7696 } 7697 7698 effect->setDevice(mDevice); 7699 effect->setMode(mAudioFlinger->getMode()); 7700 return NO_ERROR; 7701} 7702 7703void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7704 7705 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7706 effect_descriptor_t desc = effect->desc(); 7707 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7708 detachAuxEffect_l(effect->id()); 7709 } 7710 7711 sp<EffectChain> chain = effect->chain().promote(); 7712 if (chain != 0) { 7713 // remove effect chain if removing last effect 7714 if (chain->removeEffect_l(effect) == 0) { 7715 removeEffectChain_l(chain); 7716 } 7717 } else { 7718 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7719 } 7720} 7721 7722void AudioFlinger::ThreadBase::lockEffectChains_l( 7723 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7724{ 7725 effectChains = mEffectChains; 7726 for (size_t i = 0; i < mEffectChains.size(); i++) { 7727 mEffectChains[i]->lock(); 7728 } 7729} 7730 7731void AudioFlinger::ThreadBase::unlockEffectChains( 7732 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7733{ 7734 for (size_t i = 0; i < effectChains.size(); i++) { 7735 effectChains[i]->unlock(); 7736 } 7737} 7738 7739sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7740{ 7741 Mutex::Autolock _l(mLock); 7742 return getEffectChain_l(sessionId); 7743} 7744 7745sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7746{ 7747 size_t size = mEffectChains.size(); 7748 for (size_t i = 0; i < size; i++) { 7749 if (mEffectChains[i]->sessionId() == sessionId) { 7750 return mEffectChains[i]; 7751 } 7752 } 7753 return 0; 7754} 7755 7756void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7757{ 7758 Mutex::Autolock _l(mLock); 7759 size_t size = mEffectChains.size(); 7760 for (size_t i = 0; i < size; i++) { 7761 mEffectChains[i]->setMode_l(mode); 7762 } 7763} 7764 7765void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7766 const wp<EffectHandle>& handle, 7767 bool unpinIfLast) { 7768 7769 Mutex::Autolock _l(mLock); 7770 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7771 // delete the effect module if removing last handle on it 7772 if (effect->removeHandle(handle) == 0) { 7773 if (!effect->isPinned() || unpinIfLast) { 7774 removeEffect_l(effect); 7775 AudioSystem::unregisterEffect(effect->id()); 7776 } 7777 } 7778} 7779 7780status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7781{ 7782 int session = chain->sessionId(); 7783 int16_t *buffer = mMixBuffer; 7784 bool ownsBuffer = false; 7785 7786 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7787 if (session > 0) { 7788 // Only one effect chain can be present in direct output thread and it uses 7789 // the mix buffer as input 7790 if (mType != DIRECT) { 7791 size_t numSamples = mNormalFrameCount * mChannelCount; 7792 buffer = new int16_t[numSamples]; 7793 memset(buffer, 0, numSamples * sizeof(int16_t)); 7794 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7795 ownsBuffer = true; 7796 } 7797 7798 // Attach all tracks with same session ID to this chain. 7799 for (size_t i = 0; i < mTracks.size(); ++i) { 7800 sp<Track> track = mTracks[i]; 7801 if (session == track->sessionId()) { 7802 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7803 track->setMainBuffer(buffer); 7804 chain->incTrackCnt(); 7805 } 7806 } 7807 7808 // indicate all active tracks in the chain 7809 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7810 sp<Track> track = mActiveTracks[i].promote(); 7811 if (track == 0) continue; 7812 if (session == track->sessionId()) { 7813 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7814 chain->incActiveTrackCnt(); 7815 } 7816 } 7817 } 7818 7819 chain->setInBuffer(buffer, ownsBuffer); 7820 chain->setOutBuffer(mMixBuffer); 7821 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7822 // chains list in order to be processed last as it contains output stage effects 7823 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7824 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7825 // after track specific effects and before output stage 7826 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7827 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7828 // Effect chain for other sessions are inserted at beginning of effect 7829 // chains list to be processed before output mix effects. Relative order between other 7830 // sessions is not important 7831 size_t size = mEffectChains.size(); 7832 size_t i = 0; 7833 for (i = 0; i < size; i++) { 7834 if (mEffectChains[i]->sessionId() < session) break; 7835 } 7836 mEffectChains.insertAt(chain, i); 7837 checkSuspendOnAddEffectChain_l(chain); 7838 7839 return NO_ERROR; 7840} 7841 7842size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7843{ 7844 int session = chain->sessionId(); 7845 7846 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7847 7848 for (size_t i = 0; i < mEffectChains.size(); i++) { 7849 if (chain == mEffectChains[i]) { 7850 mEffectChains.removeAt(i); 7851 // detach all active tracks from the chain 7852 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7853 sp<Track> track = mActiveTracks[i].promote(); 7854 if (track == 0) continue; 7855 if (session == track->sessionId()) { 7856 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7857 chain.get(), session); 7858 chain->decActiveTrackCnt(); 7859 } 7860 } 7861 7862 // detach all tracks with same session ID from this chain 7863 for (size_t i = 0; i < mTracks.size(); ++i) { 7864 sp<Track> track = mTracks[i]; 7865 if (session == track->sessionId()) { 7866 track->setMainBuffer(mMixBuffer); 7867 chain->decTrackCnt(); 7868 } 7869 } 7870 break; 7871 } 7872 } 7873 return mEffectChains.size(); 7874} 7875 7876status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7877 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7878{ 7879 Mutex::Autolock _l(mLock); 7880 return attachAuxEffect_l(track, EffectId); 7881} 7882 7883status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7884 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7885{ 7886 status_t status = NO_ERROR; 7887 7888 if (EffectId == 0) { 7889 track->setAuxBuffer(0, NULL); 7890 } else { 7891 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7892 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7893 if (effect != 0) { 7894 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7895 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7896 } else { 7897 status = INVALID_OPERATION; 7898 } 7899 } else { 7900 status = BAD_VALUE; 7901 } 7902 } 7903 return status; 7904} 7905 7906void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7907{ 7908 for (size_t i = 0; i < mTracks.size(); ++i) { 7909 sp<Track> track = mTracks[i]; 7910 if (track->auxEffectId() == effectId) { 7911 attachAuxEffect_l(track, 0); 7912 } 7913 } 7914} 7915 7916status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7917{ 7918 // only one chain per input thread 7919 if (mEffectChains.size() != 0) { 7920 return INVALID_OPERATION; 7921 } 7922 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7923 7924 chain->setInBuffer(NULL); 7925 chain->setOutBuffer(NULL); 7926 7927 checkSuspendOnAddEffectChain_l(chain); 7928 7929 mEffectChains.add(chain); 7930 7931 return NO_ERROR; 7932} 7933 7934size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7935{ 7936 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7937 ALOGW_IF(mEffectChains.size() != 1, 7938 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7939 chain.get(), mEffectChains.size(), this); 7940 if (mEffectChains.size() == 1) { 7941 mEffectChains.removeAt(0); 7942 } 7943 return 0; 7944} 7945 7946// ---------------------------------------------------------------------------- 7947// EffectModule implementation 7948// ---------------------------------------------------------------------------- 7949 7950#undef LOG_TAG 7951#define LOG_TAG "AudioFlinger::EffectModule" 7952 7953AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7954 const wp<AudioFlinger::EffectChain>& chain, 7955 effect_descriptor_t *desc, 7956 int id, 7957 int sessionId) 7958 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7959 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7960{ 7961 ALOGV("Constructor %p", this); 7962 int lStatus; 7963 if (thread == NULL) { 7964 return; 7965 } 7966 7967 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7968 7969 // create effect engine from effect factory 7970 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7971 7972 if (mStatus != NO_ERROR) { 7973 return; 7974 } 7975 lStatus = init(); 7976 if (lStatus < 0) { 7977 mStatus = lStatus; 7978 goto Error; 7979 } 7980 7981 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7982 mPinned = true; 7983 } 7984 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7985 return; 7986Error: 7987 EffectRelease(mEffectInterface); 7988 mEffectInterface = NULL; 7989 ALOGV("Constructor Error %d", mStatus); 7990} 7991 7992AudioFlinger::EffectModule::~EffectModule() 7993{ 7994 ALOGV("Destructor %p", this); 7995 if (mEffectInterface != NULL) { 7996 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7997 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7998 sp<ThreadBase> thread = mThread.promote(); 7999 if (thread != 0) { 8000 audio_stream_t *stream = thread->stream(); 8001 if (stream != NULL) { 8002 stream->remove_audio_effect(stream, mEffectInterface); 8003 } 8004 } 8005 } 8006 // release effect engine 8007 EffectRelease(mEffectInterface); 8008 } 8009} 8010 8011status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 8012{ 8013 status_t status; 8014 8015 Mutex::Autolock _l(mLock); 8016 int priority = handle->priority(); 8017 size_t size = mHandles.size(); 8018 sp<EffectHandle> h; 8019 size_t i; 8020 for (i = 0; i < size; i++) { 8021 h = mHandles[i].promote(); 8022 if (h == 0) continue; 8023 if (h->priority() <= priority) break; 8024 } 8025 // if inserted in first place, move effect control from previous owner to this handle 8026 if (i == 0) { 8027 bool enabled = false; 8028 if (h != 0) { 8029 enabled = h->enabled(); 8030 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8031 } 8032 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8033 status = NO_ERROR; 8034 } else { 8035 status = ALREADY_EXISTS; 8036 } 8037 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8038 mHandles.insertAt(handle, i); 8039 return status; 8040} 8041 8042size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8043{ 8044 Mutex::Autolock _l(mLock); 8045 size_t size = mHandles.size(); 8046 size_t i; 8047 for (i = 0; i < size; i++) { 8048 if (mHandles[i] == handle) break; 8049 } 8050 if (i == size) { 8051 return size; 8052 } 8053 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8054 8055 bool enabled = false; 8056 EffectHandle *hdl = handle.unsafe_get(); 8057 if (hdl != NULL) { 8058 ALOGV("removeHandle() unsafe_get OK"); 8059 enabled = hdl->enabled(); 8060 } 8061 mHandles.removeAt(i); 8062 size = mHandles.size(); 8063 // if removed from first place, move effect control from this handle to next in line 8064 if (i == 0 && size != 0) { 8065 sp<EffectHandle> h = mHandles[0].promote(); 8066 if (h != 0) { 8067 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8068 } 8069 } 8070 8071 // Prevent calls to process() and other functions on effect interface from now on. 8072 // The effect engine will be released by the destructor when the last strong reference on 8073 // this object is released which can happen after next process is called. 8074 if (size == 0 && !mPinned) { 8075 mState = DESTROYED; 8076 } 8077 8078 return size; 8079} 8080 8081sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8082{ 8083 Mutex::Autolock _l(mLock); 8084 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8085} 8086 8087void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8088{ 8089 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8090 // keep a strong reference on this EffectModule to avoid calling the 8091 // destructor before we exit 8092 sp<EffectModule> keep(this); 8093 { 8094 sp<ThreadBase> thread = mThread.promote(); 8095 if (thread != 0) { 8096 thread->disconnectEffect(keep, handle, unpinIfLast); 8097 } 8098 } 8099} 8100 8101void AudioFlinger::EffectModule::updateState() { 8102 Mutex::Autolock _l(mLock); 8103 8104 switch (mState) { 8105 case RESTART: 8106 reset_l(); 8107 // FALL THROUGH 8108 8109 case STARTING: 8110 // clear auxiliary effect input buffer for next accumulation 8111 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8112 memset(mConfig.inputCfg.buffer.raw, 8113 0, 8114 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8115 } 8116 start_l(); 8117 mState = ACTIVE; 8118 break; 8119 case STOPPING: 8120 stop_l(); 8121 mDisableWaitCnt = mMaxDisableWaitCnt; 8122 mState = STOPPED; 8123 break; 8124 case STOPPED: 8125 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8126 // turn off sequence. 8127 if (--mDisableWaitCnt == 0) { 8128 reset_l(); 8129 mState = IDLE; 8130 } 8131 break; 8132 default: //IDLE , ACTIVE, DESTROYED 8133 break; 8134 } 8135} 8136 8137void AudioFlinger::EffectModule::process() 8138{ 8139 Mutex::Autolock _l(mLock); 8140 8141 if (mState == DESTROYED || mEffectInterface == NULL || 8142 mConfig.inputCfg.buffer.raw == NULL || 8143 mConfig.outputCfg.buffer.raw == NULL) { 8144 return; 8145 } 8146 8147 if (isProcessEnabled()) { 8148 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8149 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8150 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8151 mConfig.inputCfg.buffer.s32, 8152 mConfig.inputCfg.buffer.frameCount/2); 8153 } 8154 8155 // do the actual processing in the effect engine 8156 int ret = (*mEffectInterface)->process(mEffectInterface, 8157 &mConfig.inputCfg.buffer, 8158 &mConfig.outputCfg.buffer); 8159 8160 // force transition to IDLE state when engine is ready 8161 if (mState == STOPPED && ret == -ENODATA) { 8162 mDisableWaitCnt = 1; 8163 } 8164 8165 // clear auxiliary effect input buffer for next accumulation 8166 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8167 memset(mConfig.inputCfg.buffer.raw, 0, 8168 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8169 } 8170 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8171 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8172 // If an insert effect is idle and input buffer is different from output buffer, 8173 // accumulate input onto output 8174 sp<EffectChain> chain = mChain.promote(); 8175 if (chain != 0 && chain->activeTrackCnt() != 0) { 8176 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8177 int16_t *in = mConfig.inputCfg.buffer.s16; 8178 int16_t *out = mConfig.outputCfg.buffer.s16; 8179 for (size_t i = 0; i < frameCnt; i++) { 8180 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8181 } 8182 } 8183 } 8184} 8185 8186void AudioFlinger::EffectModule::reset_l() 8187{ 8188 if (mEffectInterface == NULL) { 8189 return; 8190 } 8191 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8192} 8193 8194status_t AudioFlinger::EffectModule::configure() 8195{ 8196 uint32_t channels; 8197 if (mEffectInterface == NULL) { 8198 return NO_INIT; 8199 } 8200 8201 sp<ThreadBase> thread = mThread.promote(); 8202 if (thread == 0) { 8203 return DEAD_OBJECT; 8204 } 8205 8206 // TODO: handle configuration of effects replacing track process 8207 if (thread->channelCount() == 1) { 8208 channels = AUDIO_CHANNEL_OUT_MONO; 8209 } else { 8210 channels = AUDIO_CHANNEL_OUT_STEREO; 8211 } 8212 8213 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8214 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8215 } else { 8216 mConfig.inputCfg.channels = channels; 8217 } 8218 mConfig.outputCfg.channels = channels; 8219 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8220 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8221 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8222 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8223 mConfig.inputCfg.bufferProvider.cookie = NULL; 8224 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8225 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8226 mConfig.outputCfg.bufferProvider.cookie = NULL; 8227 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8228 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8229 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8230 // Insert effect: 8231 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8232 // always overwrites output buffer: input buffer == output buffer 8233 // - in other sessions: 8234 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8235 // other effect: overwrites output buffer: input buffer == output buffer 8236 // Auxiliary effect: 8237 // accumulates in output buffer: input buffer != output buffer 8238 // Therefore: accumulate <=> input buffer != output buffer 8239 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8240 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8241 } else { 8242 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8243 } 8244 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8245 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8246 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8247 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8248 8249 ALOGV("configure() %p thread %p buffer %p framecount %d", 8250 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8251 8252 status_t cmdStatus; 8253 uint32_t size = sizeof(int); 8254 status_t status = (*mEffectInterface)->command(mEffectInterface, 8255 EFFECT_CMD_SET_CONFIG, 8256 sizeof(effect_config_t), 8257 &mConfig, 8258 &size, 8259 &cmdStatus); 8260 if (status == 0) { 8261 status = cmdStatus; 8262 } 8263 8264 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8265 (1000 * mConfig.outputCfg.buffer.frameCount); 8266 8267 return status; 8268} 8269 8270status_t AudioFlinger::EffectModule::init() 8271{ 8272 Mutex::Autolock _l(mLock); 8273 if (mEffectInterface == NULL) { 8274 return NO_INIT; 8275 } 8276 status_t cmdStatus; 8277 uint32_t size = sizeof(status_t); 8278 status_t status = (*mEffectInterface)->command(mEffectInterface, 8279 EFFECT_CMD_INIT, 8280 0, 8281 NULL, 8282 &size, 8283 &cmdStatus); 8284 if (status == 0) { 8285 status = cmdStatus; 8286 } 8287 return status; 8288} 8289 8290status_t AudioFlinger::EffectModule::start() 8291{ 8292 Mutex::Autolock _l(mLock); 8293 return start_l(); 8294} 8295 8296status_t AudioFlinger::EffectModule::start_l() 8297{ 8298 if (mEffectInterface == NULL) { 8299 return NO_INIT; 8300 } 8301 status_t cmdStatus; 8302 uint32_t size = sizeof(status_t); 8303 status_t status = (*mEffectInterface)->command(mEffectInterface, 8304 EFFECT_CMD_ENABLE, 8305 0, 8306 NULL, 8307 &size, 8308 &cmdStatus); 8309 if (status == 0) { 8310 status = cmdStatus; 8311 } 8312 if (status == 0 && 8313 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8314 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8315 sp<ThreadBase> thread = mThread.promote(); 8316 if (thread != 0) { 8317 audio_stream_t *stream = thread->stream(); 8318 if (stream != NULL) { 8319 stream->add_audio_effect(stream, mEffectInterface); 8320 } 8321 } 8322 } 8323 return status; 8324} 8325 8326status_t AudioFlinger::EffectModule::stop() 8327{ 8328 Mutex::Autolock _l(mLock); 8329 return stop_l(); 8330} 8331 8332status_t AudioFlinger::EffectModule::stop_l() 8333{ 8334 if (mEffectInterface == NULL) { 8335 return NO_INIT; 8336 } 8337 status_t cmdStatus; 8338 uint32_t size = sizeof(status_t); 8339 status_t status = (*mEffectInterface)->command(mEffectInterface, 8340 EFFECT_CMD_DISABLE, 8341 0, 8342 NULL, 8343 &size, 8344 &cmdStatus); 8345 if (status == 0) { 8346 status = cmdStatus; 8347 } 8348 if (status == 0 && 8349 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8350 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8351 sp<ThreadBase> thread = mThread.promote(); 8352 if (thread != 0) { 8353 audio_stream_t *stream = thread->stream(); 8354 if (stream != NULL) { 8355 stream->remove_audio_effect(stream, mEffectInterface); 8356 } 8357 } 8358 } 8359 return status; 8360} 8361 8362status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8363 uint32_t cmdSize, 8364 void *pCmdData, 8365 uint32_t *replySize, 8366 void *pReplyData) 8367{ 8368 Mutex::Autolock _l(mLock); 8369// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8370 8371 if (mState == DESTROYED || mEffectInterface == NULL) { 8372 return NO_INIT; 8373 } 8374 status_t status = (*mEffectInterface)->command(mEffectInterface, 8375 cmdCode, 8376 cmdSize, 8377 pCmdData, 8378 replySize, 8379 pReplyData); 8380 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8381 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8382 for (size_t i = 1; i < mHandles.size(); i++) { 8383 sp<EffectHandle> h = mHandles[i].promote(); 8384 if (h != 0) { 8385 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8386 } 8387 } 8388 } 8389 return status; 8390} 8391 8392status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8393{ 8394 8395 Mutex::Autolock _l(mLock); 8396 ALOGV("setEnabled %p enabled %d", this, enabled); 8397 8398 if (enabled != isEnabled()) { 8399 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8400 if (enabled && status != NO_ERROR) { 8401 return status; 8402 } 8403 8404 switch (mState) { 8405 // going from disabled to enabled 8406 case IDLE: 8407 mState = STARTING; 8408 break; 8409 case STOPPED: 8410 mState = RESTART; 8411 break; 8412 case STOPPING: 8413 mState = ACTIVE; 8414 break; 8415 8416 // going from enabled to disabled 8417 case RESTART: 8418 mState = STOPPED; 8419 break; 8420 case STARTING: 8421 mState = IDLE; 8422 break; 8423 case ACTIVE: 8424 mState = STOPPING; 8425 break; 8426 case DESTROYED: 8427 return NO_ERROR; // simply ignore as we are being destroyed 8428 } 8429 for (size_t i = 1; i < mHandles.size(); i++) { 8430 sp<EffectHandle> h = mHandles[i].promote(); 8431 if (h != 0) { 8432 h->setEnabled(enabled); 8433 } 8434 } 8435 } 8436 return NO_ERROR; 8437} 8438 8439bool AudioFlinger::EffectModule::isEnabled() const 8440{ 8441 switch (mState) { 8442 case RESTART: 8443 case STARTING: 8444 case ACTIVE: 8445 return true; 8446 case IDLE: 8447 case STOPPING: 8448 case STOPPED: 8449 case DESTROYED: 8450 default: 8451 return false; 8452 } 8453} 8454 8455bool AudioFlinger::EffectModule::isProcessEnabled() const 8456{ 8457 switch (mState) { 8458 case RESTART: 8459 case ACTIVE: 8460 case STOPPING: 8461 case STOPPED: 8462 return true; 8463 case IDLE: 8464 case STARTING: 8465 case DESTROYED: 8466 default: 8467 return false; 8468 } 8469} 8470 8471status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8472{ 8473 Mutex::Autolock _l(mLock); 8474 status_t status = NO_ERROR; 8475 8476 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8477 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8478 if (isProcessEnabled() && 8479 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8480 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8481 status_t cmdStatus; 8482 uint32_t volume[2]; 8483 uint32_t *pVolume = NULL; 8484 uint32_t size = sizeof(volume); 8485 volume[0] = *left; 8486 volume[1] = *right; 8487 if (controller) { 8488 pVolume = volume; 8489 } 8490 status = (*mEffectInterface)->command(mEffectInterface, 8491 EFFECT_CMD_SET_VOLUME, 8492 size, 8493 volume, 8494 &size, 8495 pVolume); 8496 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8497 *left = volume[0]; 8498 *right = volume[1]; 8499 } 8500 } 8501 return status; 8502} 8503 8504status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8505{ 8506 Mutex::Autolock _l(mLock); 8507 status_t status = NO_ERROR; 8508 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8509 // audio pre processing modules on RecordThread can receive both output and 8510 // input device indication in the same call 8511 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8512 if (dev) { 8513 status_t cmdStatus; 8514 uint32_t size = sizeof(status_t); 8515 8516 status = (*mEffectInterface)->command(mEffectInterface, 8517 EFFECT_CMD_SET_DEVICE, 8518 sizeof(uint32_t), 8519 &dev, 8520 &size, 8521 &cmdStatus); 8522 if (status == NO_ERROR) { 8523 status = cmdStatus; 8524 } 8525 } 8526 dev = device & AUDIO_DEVICE_IN_ALL; 8527 if (dev) { 8528 status_t cmdStatus; 8529 uint32_t size = sizeof(status_t); 8530 8531 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8532 EFFECT_CMD_SET_INPUT_DEVICE, 8533 sizeof(uint32_t), 8534 &dev, 8535 &size, 8536 &cmdStatus); 8537 if (status2 == NO_ERROR) { 8538 status2 = cmdStatus; 8539 } 8540 if (status == NO_ERROR) { 8541 status = status2; 8542 } 8543 } 8544 } 8545 return status; 8546} 8547 8548status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8549{ 8550 Mutex::Autolock _l(mLock); 8551 status_t status = NO_ERROR; 8552 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8553 status_t cmdStatus; 8554 uint32_t size = sizeof(status_t); 8555 status = (*mEffectInterface)->command(mEffectInterface, 8556 EFFECT_CMD_SET_AUDIO_MODE, 8557 sizeof(audio_mode_t), 8558 &mode, 8559 &size, 8560 &cmdStatus); 8561 if (status == NO_ERROR) { 8562 status = cmdStatus; 8563 } 8564 } 8565 return status; 8566} 8567 8568void AudioFlinger::EffectModule::setSuspended(bool suspended) 8569{ 8570 Mutex::Autolock _l(mLock); 8571 mSuspended = suspended; 8572} 8573 8574bool AudioFlinger::EffectModule::suspended() const 8575{ 8576 Mutex::Autolock _l(mLock); 8577 return mSuspended; 8578} 8579 8580status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8581{ 8582 const size_t SIZE = 256; 8583 char buffer[SIZE]; 8584 String8 result; 8585 8586 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8587 result.append(buffer); 8588 8589 bool locked = tryLock(mLock); 8590 // failed to lock - AudioFlinger is probably deadlocked 8591 if (!locked) { 8592 result.append("\t\tCould not lock Fx mutex:\n"); 8593 } 8594 8595 result.append("\t\tSession Status State Engine:\n"); 8596 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8597 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8598 result.append(buffer); 8599 8600 result.append("\t\tDescriptor:\n"); 8601 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8602 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8603 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8604 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8605 result.append(buffer); 8606 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8607 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8608 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8609 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8610 result.append(buffer); 8611 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8612 mDescriptor.apiVersion, 8613 mDescriptor.flags); 8614 result.append(buffer); 8615 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8616 mDescriptor.name); 8617 result.append(buffer); 8618 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8619 mDescriptor.implementor); 8620 result.append(buffer); 8621 8622 result.append("\t\t- Input configuration:\n"); 8623 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8624 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8625 (uint32_t)mConfig.inputCfg.buffer.raw, 8626 mConfig.inputCfg.buffer.frameCount, 8627 mConfig.inputCfg.samplingRate, 8628 mConfig.inputCfg.channels, 8629 mConfig.inputCfg.format); 8630 result.append(buffer); 8631 8632 result.append("\t\t- Output configuration:\n"); 8633 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8634 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8635 (uint32_t)mConfig.outputCfg.buffer.raw, 8636 mConfig.outputCfg.buffer.frameCount, 8637 mConfig.outputCfg.samplingRate, 8638 mConfig.outputCfg.channels, 8639 mConfig.outputCfg.format); 8640 result.append(buffer); 8641 8642 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8643 result.append(buffer); 8644 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8645 for (size_t i = 0; i < mHandles.size(); ++i) { 8646 sp<EffectHandle> handle = mHandles[i].promote(); 8647 if (handle != 0) { 8648 handle->dump(buffer, SIZE); 8649 result.append(buffer); 8650 } 8651 } 8652 8653 result.append("\n"); 8654 8655 write(fd, result.string(), result.length()); 8656 8657 if (locked) { 8658 mLock.unlock(); 8659 } 8660 8661 return NO_ERROR; 8662} 8663 8664// ---------------------------------------------------------------------------- 8665// EffectHandle implementation 8666// ---------------------------------------------------------------------------- 8667 8668#undef LOG_TAG 8669#define LOG_TAG "AudioFlinger::EffectHandle" 8670 8671AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8672 const sp<AudioFlinger::Client>& client, 8673 const sp<IEffectClient>& effectClient, 8674 int32_t priority) 8675 : BnEffect(), 8676 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8677 mPriority(priority), mHasControl(false), mEnabled(false) 8678{ 8679 ALOGV("constructor %p", this); 8680 8681 if (client == 0) { 8682 return; 8683 } 8684 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8685 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8686 if (mCblkMemory != 0) { 8687 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8688 8689 if (mCblk != NULL) { 8690 new(mCblk) effect_param_cblk_t(); 8691 mBuffer = (uint8_t *)mCblk + bufOffset; 8692 } 8693 } else { 8694 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8695 return; 8696 } 8697} 8698 8699AudioFlinger::EffectHandle::~EffectHandle() 8700{ 8701 ALOGV("Destructor %p", this); 8702 disconnect(false); 8703 ALOGV("Destructor DONE %p", this); 8704} 8705 8706status_t AudioFlinger::EffectHandle::enable() 8707{ 8708 ALOGV("enable %p", this); 8709 if (!mHasControl) return INVALID_OPERATION; 8710 if (mEffect == 0) return DEAD_OBJECT; 8711 8712 if (mEnabled) { 8713 return NO_ERROR; 8714 } 8715 8716 mEnabled = true; 8717 8718 sp<ThreadBase> thread = mEffect->thread().promote(); 8719 if (thread != 0) { 8720 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8721 } 8722 8723 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8724 if (mEffect->suspended()) { 8725 return NO_ERROR; 8726 } 8727 8728 status_t status = mEffect->setEnabled(true); 8729 if (status != NO_ERROR) { 8730 if (thread != 0) { 8731 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8732 } 8733 mEnabled = false; 8734 } 8735 return status; 8736} 8737 8738status_t AudioFlinger::EffectHandle::disable() 8739{ 8740 ALOGV("disable %p", this); 8741 if (!mHasControl) return INVALID_OPERATION; 8742 if (mEffect == 0) return DEAD_OBJECT; 8743 8744 if (!mEnabled) { 8745 return NO_ERROR; 8746 } 8747 mEnabled = false; 8748 8749 if (mEffect->suspended()) { 8750 return NO_ERROR; 8751 } 8752 8753 status_t status = mEffect->setEnabled(false); 8754 8755 sp<ThreadBase> thread = mEffect->thread().promote(); 8756 if (thread != 0) { 8757 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8758 } 8759 8760 return status; 8761} 8762 8763void AudioFlinger::EffectHandle::disconnect() 8764{ 8765 disconnect(true); 8766} 8767 8768void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8769{ 8770 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8771 if (mEffect == 0) { 8772 return; 8773 } 8774 mEffect->disconnect(this, unpinIfLast); 8775 8776 if (mHasControl && mEnabled) { 8777 sp<ThreadBase> thread = mEffect->thread().promote(); 8778 if (thread != 0) { 8779 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8780 } 8781 } 8782 8783 // release sp on module => module destructor can be called now 8784 mEffect.clear(); 8785 if (mClient != 0) { 8786 if (mCblk != NULL) { 8787 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8788 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8789 } 8790 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8791 // Client destructor must run with AudioFlinger mutex locked 8792 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8793 mClient.clear(); 8794 } 8795} 8796 8797status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8798 uint32_t cmdSize, 8799 void *pCmdData, 8800 uint32_t *replySize, 8801 void *pReplyData) 8802{ 8803// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8804// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8805 8806 // only get parameter command is permitted for applications not controlling the effect 8807 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8808 return INVALID_OPERATION; 8809 } 8810 if (mEffect == 0) return DEAD_OBJECT; 8811 if (mClient == 0) return INVALID_OPERATION; 8812 8813 // handle commands that are not forwarded transparently to effect engine 8814 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8815 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8816 // no risk to block the whole media server process or mixer threads is we are stuck here 8817 Mutex::Autolock _l(mCblk->lock); 8818 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8819 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8820 mCblk->serverIndex = 0; 8821 mCblk->clientIndex = 0; 8822 return BAD_VALUE; 8823 } 8824 status_t status = NO_ERROR; 8825 while (mCblk->serverIndex < mCblk->clientIndex) { 8826 int reply; 8827 uint32_t rsize = sizeof(int); 8828 int *p = (int *)(mBuffer + mCblk->serverIndex); 8829 int size = *p++; 8830 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8831 ALOGW("command(): invalid parameter block size"); 8832 break; 8833 } 8834 effect_param_t *param = (effect_param_t *)p; 8835 if (param->psize == 0 || param->vsize == 0) { 8836 ALOGW("command(): null parameter or value size"); 8837 mCblk->serverIndex += size; 8838 continue; 8839 } 8840 uint32_t psize = sizeof(effect_param_t) + 8841 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8842 param->vsize; 8843 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8844 psize, 8845 p, 8846 &rsize, 8847 &reply); 8848 // stop at first error encountered 8849 if (ret != NO_ERROR) { 8850 status = ret; 8851 *(int *)pReplyData = reply; 8852 break; 8853 } else if (reply != NO_ERROR) { 8854 *(int *)pReplyData = reply; 8855 break; 8856 } 8857 mCblk->serverIndex += size; 8858 } 8859 mCblk->serverIndex = 0; 8860 mCblk->clientIndex = 0; 8861 return status; 8862 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8863 *(int *)pReplyData = NO_ERROR; 8864 return enable(); 8865 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8866 *(int *)pReplyData = NO_ERROR; 8867 return disable(); 8868 } 8869 8870 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8871} 8872 8873void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8874{ 8875 ALOGV("setControl %p control %d", this, hasControl); 8876 8877 mHasControl = hasControl; 8878 mEnabled = enabled; 8879 8880 if (signal && mEffectClient != 0) { 8881 mEffectClient->controlStatusChanged(hasControl); 8882 } 8883} 8884 8885void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8886 uint32_t cmdSize, 8887 void *pCmdData, 8888 uint32_t replySize, 8889 void *pReplyData) 8890{ 8891 if (mEffectClient != 0) { 8892 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8893 } 8894} 8895 8896 8897 8898void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8899{ 8900 if (mEffectClient != 0) { 8901 mEffectClient->enableStatusChanged(enabled); 8902 } 8903} 8904 8905status_t AudioFlinger::EffectHandle::onTransact( 8906 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8907{ 8908 return BnEffect::onTransact(code, data, reply, flags); 8909} 8910 8911 8912void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8913{ 8914 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8915 8916 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8917 (mClient == 0) ? getpid_cached : mClient->pid(), 8918 mPriority, 8919 mHasControl, 8920 !locked, 8921 mCblk ? mCblk->clientIndex : 0, 8922 mCblk ? mCblk->serverIndex : 0 8923 ); 8924 8925 if (locked) { 8926 mCblk->lock.unlock(); 8927 } 8928} 8929 8930#undef LOG_TAG 8931#define LOG_TAG "AudioFlinger::EffectChain" 8932 8933AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8934 int sessionId) 8935 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8936 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8937 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8938{ 8939 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8940 if (thread == NULL) { 8941 return; 8942 } 8943 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8944 thread->frameCount(); 8945} 8946 8947AudioFlinger::EffectChain::~EffectChain() 8948{ 8949 if (mOwnInBuffer) { 8950 delete mInBuffer; 8951 } 8952 8953} 8954 8955// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8956sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8957{ 8958 size_t size = mEffects.size(); 8959 8960 for (size_t i = 0; i < size; i++) { 8961 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8962 return mEffects[i]; 8963 } 8964 } 8965 return 0; 8966} 8967 8968// getEffectFromId_l() must be called with ThreadBase::mLock held 8969sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8970{ 8971 size_t size = mEffects.size(); 8972 8973 for (size_t i = 0; i < size; i++) { 8974 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8975 if (id == 0 || mEffects[i]->id() == id) { 8976 return mEffects[i]; 8977 } 8978 } 8979 return 0; 8980} 8981 8982// getEffectFromType_l() must be called with ThreadBase::mLock held 8983sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8984 const effect_uuid_t *type) 8985{ 8986 size_t size = mEffects.size(); 8987 8988 for (size_t i = 0; i < size; i++) { 8989 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8990 return mEffects[i]; 8991 } 8992 } 8993 return 0; 8994} 8995 8996void AudioFlinger::EffectChain::clearInputBuffer() 8997{ 8998 Mutex::Autolock _l(mLock); 8999 sp<ThreadBase> thread = mThread.promote(); 9000 if (thread == 0) { 9001 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9002 return; 9003 } 9004 clearInputBuffer_l(thread); 9005} 9006 9007// Must be called with EffectChain::mLock locked 9008void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9009{ 9010 size_t numSamples = thread->frameCount() * thread->channelCount(); 9011 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9012 9013} 9014 9015// Must be called with EffectChain::mLock locked 9016void AudioFlinger::EffectChain::process_l() 9017{ 9018 sp<ThreadBase> thread = mThread.promote(); 9019 if (thread == 0) { 9020 ALOGW("process_l(): cannot promote mixer thread"); 9021 return; 9022 } 9023 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9024 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9025 // always process effects unless no more tracks are on the session and the effect tail 9026 // has been rendered 9027 bool doProcess = true; 9028 if (!isGlobalSession) { 9029 bool tracksOnSession = (trackCnt() != 0); 9030 9031 if (!tracksOnSession && mTailBufferCount == 0) { 9032 doProcess = false; 9033 } 9034 9035 if (activeTrackCnt() == 0) { 9036 // if no track is active and the effect tail has not been rendered, 9037 // the input buffer must be cleared here as the mixer process will not do it 9038 if (tracksOnSession || mTailBufferCount > 0) { 9039 clearInputBuffer_l(thread); 9040 if (mTailBufferCount > 0) { 9041 mTailBufferCount--; 9042 } 9043 } 9044 } 9045 } 9046 9047 size_t size = mEffects.size(); 9048 if (doProcess) { 9049 for (size_t i = 0; i < size; i++) { 9050 mEffects[i]->process(); 9051 } 9052 } 9053 for (size_t i = 0; i < size; i++) { 9054 mEffects[i]->updateState(); 9055 } 9056} 9057 9058// addEffect_l() must be called with PlaybackThread::mLock held 9059status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9060{ 9061 effect_descriptor_t desc = effect->desc(); 9062 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9063 9064 Mutex::Autolock _l(mLock); 9065 effect->setChain(this); 9066 sp<ThreadBase> thread = mThread.promote(); 9067 if (thread == 0) { 9068 return NO_INIT; 9069 } 9070 effect->setThread(thread); 9071 9072 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9073 // Auxiliary effects are inserted at the beginning of mEffects vector as 9074 // they are processed first and accumulated in chain input buffer 9075 mEffects.insertAt(effect, 0); 9076 9077 // the input buffer for auxiliary effect contains mono samples in 9078 // 32 bit format. This is to avoid saturation in AudoMixer 9079 // accumulation stage. Saturation is done in EffectModule::process() before 9080 // calling the process in effect engine 9081 size_t numSamples = thread->frameCount(); 9082 int32_t *buffer = new int32_t[numSamples]; 9083 memset(buffer, 0, numSamples * sizeof(int32_t)); 9084 effect->setInBuffer((int16_t *)buffer); 9085 // auxiliary effects output samples to chain input buffer for further processing 9086 // by insert effects 9087 effect->setOutBuffer(mInBuffer); 9088 } else { 9089 // Insert effects are inserted at the end of mEffects vector as they are processed 9090 // after track and auxiliary effects. 9091 // Insert effect order as a function of indicated preference: 9092 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9093 // another effect is present 9094 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9095 // last effect claiming first position 9096 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9097 // first effect claiming last position 9098 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9099 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9100 // already present 9101 9102 size_t size = mEffects.size(); 9103 size_t idx_insert = size; 9104 ssize_t idx_insert_first = -1; 9105 ssize_t idx_insert_last = -1; 9106 9107 for (size_t i = 0; i < size; i++) { 9108 effect_descriptor_t d = mEffects[i]->desc(); 9109 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9110 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9111 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9112 // check invalid effect chaining combinations 9113 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9114 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9115 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9116 return INVALID_OPERATION; 9117 } 9118 // remember position of first insert effect and by default 9119 // select this as insert position for new effect 9120 if (idx_insert == size) { 9121 idx_insert = i; 9122 } 9123 // remember position of last insert effect claiming 9124 // first position 9125 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9126 idx_insert_first = i; 9127 } 9128 // remember position of first insert effect claiming 9129 // last position 9130 if (iPref == EFFECT_FLAG_INSERT_LAST && 9131 idx_insert_last == -1) { 9132 idx_insert_last = i; 9133 } 9134 } 9135 } 9136 9137 // modify idx_insert from first position if needed 9138 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9139 if (idx_insert_last != -1) { 9140 idx_insert = idx_insert_last; 9141 } else { 9142 idx_insert = size; 9143 } 9144 } else { 9145 if (idx_insert_first != -1) { 9146 idx_insert = idx_insert_first + 1; 9147 } 9148 } 9149 9150 // always read samples from chain input buffer 9151 effect->setInBuffer(mInBuffer); 9152 9153 // if last effect in the chain, output samples to chain 9154 // output buffer, otherwise to chain input buffer 9155 if (idx_insert == size) { 9156 if (idx_insert != 0) { 9157 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9158 mEffects[idx_insert-1]->configure(); 9159 } 9160 effect->setOutBuffer(mOutBuffer); 9161 } else { 9162 effect->setOutBuffer(mInBuffer); 9163 } 9164 mEffects.insertAt(effect, idx_insert); 9165 9166 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9167 } 9168 effect->configure(); 9169 return NO_ERROR; 9170} 9171 9172// removeEffect_l() must be called with PlaybackThread::mLock held 9173size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9174{ 9175 Mutex::Autolock _l(mLock); 9176 size_t size = mEffects.size(); 9177 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9178 9179 for (size_t i = 0; i < size; i++) { 9180 if (effect == mEffects[i]) { 9181 // calling stop here will remove pre-processing effect from the audio HAL. 9182 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9183 // the middle of a read from audio HAL 9184 if (mEffects[i]->state() == EffectModule::ACTIVE || 9185 mEffects[i]->state() == EffectModule::STOPPING) { 9186 mEffects[i]->stop(); 9187 } 9188 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9189 delete[] effect->inBuffer(); 9190 } else { 9191 if (i == size - 1 && i != 0) { 9192 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9193 mEffects[i - 1]->configure(); 9194 } 9195 } 9196 mEffects.removeAt(i); 9197 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9198 break; 9199 } 9200 } 9201 9202 return mEffects.size(); 9203} 9204 9205// setDevice_l() must be called with PlaybackThread::mLock held 9206void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9207{ 9208 size_t size = mEffects.size(); 9209 for (size_t i = 0; i < size; i++) { 9210 mEffects[i]->setDevice(device); 9211 } 9212} 9213 9214// setMode_l() must be called with PlaybackThread::mLock held 9215void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9216{ 9217 size_t size = mEffects.size(); 9218 for (size_t i = 0; i < size; i++) { 9219 mEffects[i]->setMode(mode); 9220 } 9221} 9222 9223// setVolume_l() must be called with PlaybackThread::mLock held 9224bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9225{ 9226 uint32_t newLeft = *left; 9227 uint32_t newRight = *right; 9228 bool hasControl = false; 9229 int ctrlIdx = -1; 9230 size_t size = mEffects.size(); 9231 9232 // first update volume controller 9233 for (size_t i = size; i > 0; i--) { 9234 if (mEffects[i - 1]->isProcessEnabled() && 9235 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9236 ctrlIdx = i - 1; 9237 hasControl = true; 9238 break; 9239 } 9240 } 9241 9242 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9243 if (hasControl) { 9244 *left = mNewLeftVolume; 9245 *right = mNewRightVolume; 9246 } 9247 return hasControl; 9248 } 9249 9250 mVolumeCtrlIdx = ctrlIdx; 9251 mLeftVolume = newLeft; 9252 mRightVolume = newRight; 9253 9254 // second get volume update from volume controller 9255 if (ctrlIdx >= 0) { 9256 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9257 mNewLeftVolume = newLeft; 9258 mNewRightVolume = newRight; 9259 } 9260 // then indicate volume to all other effects in chain. 9261 // Pass altered volume to effects before volume controller 9262 // and requested volume to effects after controller 9263 uint32_t lVol = newLeft; 9264 uint32_t rVol = newRight; 9265 9266 for (size_t i = 0; i < size; i++) { 9267 if ((int)i == ctrlIdx) continue; 9268 // this also works for ctrlIdx == -1 when there is no volume controller 9269 if ((int)i > ctrlIdx) { 9270 lVol = *left; 9271 rVol = *right; 9272 } 9273 mEffects[i]->setVolume(&lVol, &rVol, false); 9274 } 9275 *left = newLeft; 9276 *right = newRight; 9277 9278 return hasControl; 9279} 9280 9281status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9282{ 9283 const size_t SIZE = 256; 9284 char buffer[SIZE]; 9285 String8 result; 9286 9287 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9288 result.append(buffer); 9289 9290 bool locked = tryLock(mLock); 9291 // failed to lock - AudioFlinger is probably deadlocked 9292 if (!locked) { 9293 result.append("\tCould not lock mutex:\n"); 9294 } 9295 9296 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9297 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9298 mEffects.size(), 9299 (uint32_t)mInBuffer, 9300 (uint32_t)mOutBuffer, 9301 mActiveTrackCnt); 9302 result.append(buffer); 9303 write(fd, result.string(), result.size()); 9304 9305 for (size_t i = 0; i < mEffects.size(); ++i) { 9306 sp<EffectModule> effect = mEffects[i]; 9307 if (effect != 0) { 9308 effect->dump(fd, args); 9309 } 9310 } 9311 9312 if (locked) { 9313 mLock.unlock(); 9314 } 9315 9316 return NO_ERROR; 9317} 9318 9319// must be called with ThreadBase::mLock held 9320void AudioFlinger::EffectChain::setEffectSuspended_l( 9321 const effect_uuid_t *type, bool suspend) 9322{ 9323 sp<SuspendedEffectDesc> desc; 9324 // use effect type UUID timelow as key as there is no real risk of identical 9325 // timeLow fields among effect type UUIDs. 9326 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9327 if (suspend) { 9328 if (index >= 0) { 9329 desc = mSuspendedEffects.valueAt(index); 9330 } else { 9331 desc = new SuspendedEffectDesc(); 9332 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9333 mSuspendedEffects.add(type->timeLow, desc); 9334 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9335 } 9336 if (desc->mRefCount++ == 0) { 9337 sp<EffectModule> effect = getEffectIfEnabled(type); 9338 if (effect != 0) { 9339 desc->mEffect = effect; 9340 effect->setSuspended(true); 9341 effect->setEnabled(false); 9342 } 9343 } 9344 } else { 9345 if (index < 0) { 9346 return; 9347 } 9348 desc = mSuspendedEffects.valueAt(index); 9349 if (desc->mRefCount <= 0) { 9350 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9351 desc->mRefCount = 1; 9352 } 9353 if (--desc->mRefCount == 0) { 9354 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9355 if (desc->mEffect != 0) { 9356 sp<EffectModule> effect = desc->mEffect.promote(); 9357 if (effect != 0) { 9358 effect->setSuspended(false); 9359 sp<EffectHandle> handle = effect->controlHandle(); 9360 if (handle != 0) { 9361 effect->setEnabled(handle->enabled()); 9362 } 9363 } 9364 desc->mEffect.clear(); 9365 } 9366 mSuspendedEffects.removeItemsAt(index); 9367 } 9368 } 9369} 9370 9371// must be called with ThreadBase::mLock held 9372void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9373{ 9374 sp<SuspendedEffectDesc> desc; 9375 9376 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9377 if (suspend) { 9378 if (index >= 0) { 9379 desc = mSuspendedEffects.valueAt(index); 9380 } else { 9381 desc = new SuspendedEffectDesc(); 9382 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9383 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9384 } 9385 if (desc->mRefCount++ == 0) { 9386 Vector< sp<EffectModule> > effects; 9387 getSuspendEligibleEffects(effects); 9388 for (size_t i = 0; i < effects.size(); i++) { 9389 setEffectSuspended_l(&effects[i]->desc().type, true); 9390 } 9391 } 9392 } else { 9393 if (index < 0) { 9394 return; 9395 } 9396 desc = mSuspendedEffects.valueAt(index); 9397 if (desc->mRefCount <= 0) { 9398 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9399 desc->mRefCount = 1; 9400 } 9401 if (--desc->mRefCount == 0) { 9402 Vector<const effect_uuid_t *> types; 9403 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9404 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9405 continue; 9406 } 9407 types.add(&mSuspendedEffects.valueAt(i)->mType); 9408 } 9409 for (size_t i = 0; i < types.size(); i++) { 9410 setEffectSuspended_l(types[i], false); 9411 } 9412 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9413 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9414 } 9415 } 9416} 9417 9418 9419// The volume effect is used for automated tests only 9420#ifndef OPENSL_ES_H_ 9421static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9422 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9423const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9424#endif //OPENSL_ES_H_ 9425 9426bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9427{ 9428 // auxiliary effects and visualizer are never suspended on output mix 9429 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9430 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9431 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9432 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9433 return false; 9434 } 9435 return true; 9436} 9437 9438void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9439{ 9440 effects.clear(); 9441 for (size_t i = 0; i < mEffects.size(); i++) { 9442 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9443 effects.add(mEffects[i]); 9444 } 9445 } 9446} 9447 9448sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9449 const effect_uuid_t *type) 9450{ 9451 sp<EffectModule> effect = getEffectFromType_l(type); 9452 return effect != 0 && effect->isEnabled() ? effect : 0; 9453} 9454 9455void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9456 bool enabled) 9457{ 9458 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9459 if (enabled) { 9460 if (index < 0) { 9461 // if the effect is not suspend check if all effects are suspended 9462 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9463 if (index < 0) { 9464 return; 9465 } 9466 if (!isEffectEligibleForSuspend(effect->desc())) { 9467 return; 9468 } 9469 setEffectSuspended_l(&effect->desc().type, enabled); 9470 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9471 if (index < 0) { 9472 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9473 return; 9474 } 9475 } 9476 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9477 effect->desc().type.timeLow); 9478 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9479 // if effect is requested to suspended but was not yet enabled, supend it now. 9480 if (desc->mEffect == 0) { 9481 desc->mEffect = effect; 9482 effect->setEnabled(false); 9483 effect->setSuspended(true); 9484 } 9485 } else { 9486 if (index < 0) { 9487 return; 9488 } 9489 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9490 effect->desc().type.timeLow); 9491 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9492 desc->mEffect.clear(); 9493 effect->setSuspended(false); 9494 } 9495} 9496 9497#undef LOG_TAG 9498#define LOG_TAG "AudioFlinger" 9499 9500// ---------------------------------------------------------------------------- 9501 9502status_t AudioFlinger::onTransact( 9503 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9504{ 9505 return BnAudioFlinger::onTransact(code, data, reply, flags); 9506} 9507 9508}; // namespace android 9509