AudioFlinger.cpp revision 6d80297a55ab12759ee00b7f99fa97584b430da0
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#ifdef HAVE_REQUEST_PRIORITY 87#include "SchedulingPolicyService.h" 88#endif 89 90#ifdef SOAKER 91#include "Soaker.h" 92#endif 93 94// ---------------------------------------------------------------------------- 95 96// Note: the following macro is used for extremely verbose logging message. In 97// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 98// 0; but one side effect of this is to turn all LOGV's as well. Some messages 99// are so verbose that we want to suppress them even when we have ALOG_ASSERT 100// turned on. Do not uncomment the #def below unless you really know what you 101// are doing and want to see all of the extremely verbose messages. 102//#define VERY_VERY_VERBOSE_LOGGING 103#ifdef VERY_VERY_VERBOSE_LOGGING 104#define ALOGVV ALOGV 105#else 106#define ALOGVV(a...) do { } while(0) 107#endif 108 109namespace android { 110 111static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 112static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 113 114static const float MAX_GAIN = 4096.0f; 115static const uint32_t MAX_GAIN_INT = 0x1000; 116 117// retry counts for buffer fill timeout 118// 50 * ~20msecs = 1 second 119static const int8_t kMaxTrackRetries = 50; 120static const int8_t kMaxTrackStartupRetries = 50; 121// allow less retry attempts on direct output thread. 122// direct outputs can be a scarce resource in audio hardware and should 123// be released as quickly as possible. 124static const int8_t kMaxTrackRetriesDirect = 2; 125 126static const int kDumpLockRetries = 50; 127static const int kDumpLockSleepUs = 20000; 128 129// don't warn about blocked writes or record buffer overflows more often than this 130static const nsecs_t kWarningThrottleNs = seconds(5); 131 132// RecordThread loop sleep time upon application overrun or audio HAL read error 133static const int kRecordThreadSleepUs = 5000; 134 135// maximum time to wait for setParameters to complete 136static const nsecs_t kSetParametersTimeoutNs = seconds(2); 137 138// minimum sleep time for the mixer thread loop when tracks are active but in underrun 139static const uint32_t kMinThreadSleepTimeUs = 5000; 140// maximum divider applied to the active sleep time in the mixer thread loop 141static const uint32_t kMaxThreadSleepTimeShift = 2; 142 143// minimum normal mix buffer size, expressed in milliseconds rather than frames 144static const uint32_t kMinNormalMixBufferSizeMs = 20; 145// maximum normal mix buffer size 146static const uint32_t kMaxNormalMixBufferSizeMs = 24; 147 148nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 149 150// Whether to use fast mixer 151static const enum { 152 FastMixer_Never, // never initialize or use: for debugging only 153 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 154 // normal mixer multiplier is 1 155 FastMixer_Static, // initialize if needed, then use all the time if initialized, 156 // multiplier is calculated based on min & max normal mixer buffer size 157 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 158 // multiplier is calculated based on min & max normal mixer buffer size 159 // FIXME for FastMixer_Dynamic: 160 // Supporting this option will require fixing HALs that can't handle large writes. 161 // For example, one HAL implementation returns an error from a large write, 162 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 163 // We could either fix the HAL implementations, or provide a wrapper that breaks 164 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 165} kUseFastMixer = FastMixer_Static; 166 167// ---------------------------------------------------------------------------- 168 169#ifdef ADD_BATTERY_DATA 170// To collect the amplifier usage 171static void addBatteryData(uint32_t params) { 172 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 173 if (service == NULL) { 174 // it already logged 175 return; 176 } 177 178 service->addBatteryData(params); 179} 180#endif 181 182static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 183{ 184 const hw_module_t *mod; 185 int rc; 186 187 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 188 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 189 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 190 if (rc) { 191 goto out; 192 } 193 rc = audio_hw_device_open(mod, dev); 194 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 195 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 196 if (rc) { 197 goto out; 198 } 199 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 200 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 201 rc = BAD_VALUE; 202 goto out; 203 } 204 return 0; 205 206out: 207 *dev = NULL; 208 return rc; 209} 210 211// ---------------------------------------------------------------------------- 212 213AudioFlinger::AudioFlinger() 214 : BnAudioFlinger(), 215 mPrimaryHardwareDev(NULL), 216 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 217 mMasterVolume(1.0f), 218 mMasterVolumeSupportLvl(MVS_NONE), 219 mMasterMute(false), 220 mNextUniqueId(1), 221 mMode(AUDIO_MODE_INVALID), 222 mBtNrecIsOff(false) 223{ 224} 225 226void AudioFlinger::onFirstRef() 227{ 228 int rc = 0; 229 230 Mutex::Autolock _l(mLock); 231 232 /* TODO: move all this work into an Init() function */ 233 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 234 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 235 uint32_t int_val; 236 if (1 == sscanf(val_str, "%u", &int_val)) { 237 mStandbyTimeInNsecs = milliseconds(int_val); 238 ALOGI("Using %u mSec as standby time.", int_val); 239 } else { 240 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 241 ALOGI("Using default %u mSec as standby time.", 242 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 243 } 244 } 245 246 mMode = AUDIO_MODE_NORMAL; 247 mMasterVolumeSW = 1.0; 248 mMasterVolume = 1.0; 249 mHardwareStatus = AUDIO_HW_IDLE; 250} 251 252AudioFlinger::~AudioFlinger() 253{ 254 255 while (!mRecordThreads.isEmpty()) { 256 // closeInput() will remove first entry from mRecordThreads 257 closeInput(mRecordThreads.keyAt(0)); 258 } 259 while (!mPlaybackThreads.isEmpty()) { 260 // closeOutput() will remove first entry from mPlaybackThreads 261 closeOutput(mPlaybackThreads.keyAt(0)); 262 } 263 264 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 265 // no mHardwareLock needed, as there are no other references to this 266 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 267 delete mAudioHwDevs.valueAt(i); 268 } 269} 270 271static const char * const audio_interfaces[] = { 272 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 273 AUDIO_HARDWARE_MODULE_ID_A2DP, 274 AUDIO_HARDWARE_MODULE_ID_USB, 275}; 276#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 277 278audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 279{ 280 // if module is 0, the request comes from an old policy manager and we should load 281 // well known modules 282 if (module == 0) { 283 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 284 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 285 loadHwModule_l(audio_interfaces[i]); 286 } 287 } else { 288 // check a match for the requested module handle 289 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 290 if (audioHwdevice != NULL) { 291 return audioHwdevice->hwDevice(); 292 } 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 297 if ((dev->get_supported_devices(dev) & devices) == devices) 298 return dev; 299 } 300 301 return NULL; 302} 303 304status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Global session refs:\n"); 320 result.append(" session pid count\n"); 321 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 322 AudioSessionRef *r = mAudioSessionRefs[i]; 323 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 324 result.append(buffer); 325 } 326 write(fd, result.string(), result.size()); 327 return NO_ERROR; 328} 329 330 331status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 332{ 333 const size_t SIZE = 256; 334 char buffer[SIZE]; 335 String8 result; 336 hardware_call_state hardwareStatus = mHardwareStatus; 337 338 snprintf(buffer, SIZE, "Hardware status: %d\n" 339 "Standby Time mSec: %u\n", 340 hardwareStatus, 341 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 342 result.append(buffer); 343 write(fd, result.string(), result.size()); 344 return NO_ERROR; 345} 346 347status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 348{ 349 const size_t SIZE = 256; 350 char buffer[SIZE]; 351 String8 result; 352 snprintf(buffer, SIZE, "Permission Denial: " 353 "can't dump AudioFlinger from pid=%d, uid=%d\n", 354 IPCThreadState::self()->getCallingPid(), 355 IPCThreadState::self()->getCallingUid()); 356 result.append(buffer); 357 write(fd, result.string(), result.size()); 358 return NO_ERROR; 359} 360 361static bool tryLock(Mutex& mutex) 362{ 363 bool locked = false; 364 for (int i = 0; i < kDumpLockRetries; ++i) { 365 if (mutex.tryLock() == NO_ERROR) { 366 locked = true; 367 break; 368 } 369 usleep(kDumpLockSleepUs); 370 } 371 return locked; 372} 373 374status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 375{ 376 if (!dumpAllowed()) { 377 dumpPermissionDenial(fd, args); 378 } else { 379 // get state of hardware lock 380 bool hardwareLocked = tryLock(mHardwareLock); 381 if (!hardwareLocked) { 382 String8 result(kHardwareLockedString); 383 write(fd, result.string(), result.size()); 384 } else { 385 mHardwareLock.unlock(); 386 } 387 388 bool locked = tryLock(mLock); 389 390 // failed to lock - AudioFlinger is probably deadlocked 391 if (!locked) { 392 String8 result(kDeadlockedString); 393 write(fd, result.string(), result.size()); 394 } 395 396 dumpClients(fd, args); 397 dumpInternals(fd, args); 398 399 // dump playback threads 400 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 401 mPlaybackThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump record threads 405 for (size_t i = 0; i < mRecordThreads.size(); i++) { 406 mRecordThreads.valueAt(i)->dump(fd, args); 407 } 408 409 // dump all hardware devs 410 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 411 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 412 dev->dump(dev, fd); 413 } 414 if (locked) mLock.unlock(); 415 } 416 return NO_ERROR; 417} 418 419sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 420{ 421 // If pid is already in the mClients wp<> map, then use that entry 422 // (for which promote() is always != 0), otherwise create a new entry and Client. 423 sp<Client> client = mClients.valueFor(pid).promote(); 424 if (client == 0) { 425 client = new Client(this, pid); 426 mClients.add(pid, client); 427 } 428 429 return client; 430} 431 432// IAudioFlinger interface 433 434 435sp<IAudioTrack> AudioFlinger::createTrack( 436 pid_t pid, 437 audio_stream_type_t streamType, 438 uint32_t sampleRate, 439 audio_format_t format, 440 uint32_t channelMask, 441 int frameCount, 442 IAudioFlinger::track_flags_t flags, 443 const sp<IMemory>& sharedBuffer, 444 audio_io_handle_t output, 445 pid_t tid, 446 int *sessionId, 447 status_t *status) 448{ 449 sp<PlaybackThread::Track> track; 450 sp<TrackHandle> trackHandle; 451 sp<Client> client; 452 status_t lStatus; 453 int lSessionId; 454 455 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 456 // but if someone uses binder directly they could bypass that and cause us to crash 457 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 458 ALOGE("createTrack() invalid stream type %d", streamType); 459 lStatus = BAD_VALUE; 460 goto Exit; 461 } 462 463 { 464 Mutex::Autolock _l(mLock); 465 PlaybackThread *thread = checkPlaybackThread_l(output); 466 PlaybackThread *effectThread = NULL; 467 if (thread == NULL) { 468 ALOGE("unknown output thread"); 469 lStatus = BAD_VALUE; 470 goto Exit; 471 } 472 473 client = registerPid_l(pid); 474 475 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 476 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 477 // check if an effect chain with the same session ID is present on another 478 // output thread and move it here. 479 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 480 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 481 if (mPlaybackThreads.keyAt(i) != output) { 482 uint32_t sessions = t->hasAudioSession(*sessionId); 483 if (sessions & PlaybackThread::EFFECT_SESSION) { 484 effectThread = t.get(); 485 break; 486 } 487 } 488 } 489 lSessionId = *sessionId; 490 } else { 491 // if no audio session id is provided, create one here 492 lSessionId = nextUniqueId(); 493 if (sessionId != NULL) { 494 *sessionId = lSessionId; 495 } 496 } 497 ALOGV("createTrack() lSessionId: %d", lSessionId); 498 499 track = thread->createTrack_l(client, streamType, sampleRate, format, 500 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 501 502 // move effect chain to this output thread if an effect on same session was waiting 503 // for a track to be created 504 if (lStatus == NO_ERROR && effectThread != NULL) { 505 Mutex::Autolock _dl(thread->mLock); 506 Mutex::Autolock _sl(effectThread->mLock); 507 moveEffectChain_l(lSessionId, effectThread, thread, true); 508 } 509 510 // Look for sync events awaiting for a session to be used. 511 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 512 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 513 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 514 if (lStatus == NO_ERROR) { 515 track->setSyncEvent(mPendingSyncEvents[i]); 516 } else { 517 mPendingSyncEvents[i]->cancel(); 518 } 519 mPendingSyncEvents.removeAt(i); 520 i--; 521 } 522 } 523 } 524 } 525 if (lStatus == NO_ERROR) { 526 trackHandle = new TrackHandle(track); 527 } else { 528 // remove local strong reference to Client before deleting the Track so that the Client 529 // destructor is called by the TrackBase destructor with mLock held 530 client.clear(); 531 track.clear(); 532 } 533 534Exit: 535 if (status != NULL) { 536 *status = lStatus; 537 } 538 return trackHandle; 539} 540 541uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 542{ 543 Mutex::Autolock _l(mLock); 544 PlaybackThread *thread = checkPlaybackThread_l(output); 545 if (thread == NULL) { 546 ALOGW("sampleRate() unknown thread %d", output); 547 return 0; 548 } 549 return thread->sampleRate(); 550} 551 552int AudioFlinger::channelCount(audio_io_handle_t output) const 553{ 554 Mutex::Autolock _l(mLock); 555 PlaybackThread *thread = checkPlaybackThread_l(output); 556 if (thread == NULL) { 557 ALOGW("channelCount() unknown thread %d", output); 558 return 0; 559 } 560 return thread->channelCount(); 561} 562 563audio_format_t AudioFlinger::format(audio_io_handle_t output) const 564{ 565 Mutex::Autolock _l(mLock); 566 PlaybackThread *thread = checkPlaybackThread_l(output); 567 if (thread == NULL) { 568 ALOGW("format() unknown thread %d", output); 569 return AUDIO_FORMAT_INVALID; 570 } 571 return thread->format(); 572} 573 574size_t AudioFlinger::frameCount(audio_io_handle_t output) const 575{ 576 Mutex::Autolock _l(mLock); 577 PlaybackThread *thread = checkPlaybackThread_l(output); 578 if (thread == NULL) { 579 ALOGW("frameCount() unknown thread %d", output); 580 return 0; 581 } 582 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 583 // should examine all callers and fix them to handle smaller counts 584 return thread->frameCount(); 585} 586 587uint32_t AudioFlinger::latency(audio_io_handle_t output) const 588{ 589 Mutex::Autolock _l(mLock); 590 PlaybackThread *thread = checkPlaybackThread_l(output); 591 if (thread == NULL) { 592 ALOGW("latency() unknown thread %d", output); 593 return 0; 594 } 595 return thread->latency(); 596} 597 598status_t AudioFlinger::setMasterVolume(float value) 599{ 600 status_t ret = initCheck(); 601 if (ret != NO_ERROR) { 602 return ret; 603 } 604 605 // check calling permissions 606 if (!settingsAllowed()) { 607 return PERMISSION_DENIED; 608 } 609 610 float swmv = value; 611 612 Mutex::Autolock _l(mLock); 613 614 // when hw supports master volume, don't scale in sw mixer 615 if (MVS_NONE != mMasterVolumeSupportLvl) { 616 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 617 AutoMutex lock(mHardwareLock); 618 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 619 620 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 621 if (NULL != dev->set_master_volume) { 622 dev->set_master_volume(dev, value); 623 } 624 mHardwareStatus = AUDIO_HW_IDLE; 625 } 626 627 swmv = 1.0; 628 } 629 630 mMasterVolume = value; 631 mMasterVolumeSW = swmv; 632 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 633 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 634 635 return NO_ERROR; 636} 637 638status_t AudioFlinger::setMode(audio_mode_t mode) 639{ 640 status_t ret = initCheck(); 641 if (ret != NO_ERROR) { 642 return ret; 643 } 644 645 // check calling permissions 646 if (!settingsAllowed()) { 647 return PERMISSION_DENIED; 648 } 649 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 650 ALOGW("Illegal value: setMode(%d)", mode); 651 return BAD_VALUE; 652 } 653 654 { // scope for the lock 655 AutoMutex lock(mHardwareLock); 656 mHardwareStatus = AUDIO_HW_SET_MODE; 657 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 658 mHardwareStatus = AUDIO_HW_IDLE; 659 } 660 661 if (NO_ERROR == ret) { 662 Mutex::Autolock _l(mLock); 663 mMode = mode; 664 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 665 mPlaybackThreads.valueAt(i)->setMode(mode); 666 } 667 668 return ret; 669} 670 671status_t AudioFlinger::setMicMute(bool state) 672{ 673 status_t ret = initCheck(); 674 if (ret != NO_ERROR) { 675 return ret; 676 } 677 678 // check calling permissions 679 if (!settingsAllowed()) { 680 return PERMISSION_DENIED; 681 } 682 683 AutoMutex lock(mHardwareLock); 684 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 685 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 686 mHardwareStatus = AUDIO_HW_IDLE; 687 return ret; 688} 689 690bool AudioFlinger::getMicMute() const 691{ 692 status_t ret = initCheck(); 693 if (ret != NO_ERROR) { 694 return false; 695 } 696 697 bool state = AUDIO_MODE_INVALID; 698 AutoMutex lock(mHardwareLock); 699 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 700 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 701 mHardwareStatus = AUDIO_HW_IDLE; 702 return state; 703} 704 705status_t AudioFlinger::setMasterMute(bool muted) 706{ 707 // check calling permissions 708 if (!settingsAllowed()) { 709 return PERMISSION_DENIED; 710 } 711 712 Mutex::Autolock _l(mLock); 713 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 714 mMasterMute = muted; 715 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 716 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 717 718 return NO_ERROR; 719} 720 721float AudioFlinger::masterVolume() const 722{ 723 Mutex::Autolock _l(mLock); 724 return masterVolume_l(); 725} 726 727float AudioFlinger::masterVolumeSW() const 728{ 729 Mutex::Autolock _l(mLock); 730 return masterVolumeSW_l(); 731} 732 733bool AudioFlinger::masterMute() const 734{ 735 Mutex::Autolock _l(mLock); 736 return masterMute_l(); 737} 738 739float AudioFlinger::masterVolume_l() const 740{ 741 if (MVS_FULL == mMasterVolumeSupportLvl) { 742 float ret_val; 743 AutoMutex lock(mHardwareLock); 744 745 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 746 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 747 (NULL != mPrimaryHardwareDev->get_master_volume), 748 "can't get master volume"); 749 750 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 751 mHardwareStatus = AUDIO_HW_IDLE; 752 return ret_val; 753 } 754 755 return mMasterVolume; 756} 757 758status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 759 audio_io_handle_t output) 760{ 761 // check calling permissions 762 if (!settingsAllowed()) { 763 return PERMISSION_DENIED; 764 } 765 766 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 767 ALOGE("setStreamVolume() invalid stream %d", stream); 768 return BAD_VALUE; 769 } 770 771 AutoMutex lock(mLock); 772 PlaybackThread *thread = NULL; 773 if (output) { 774 thread = checkPlaybackThread_l(output); 775 if (thread == NULL) { 776 return BAD_VALUE; 777 } 778 } 779 780 mStreamTypes[stream].volume = value; 781 782 if (thread == NULL) { 783 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 784 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 785 } 786 } else { 787 thread->setStreamVolume(stream, value); 788 } 789 790 return NO_ERROR; 791} 792 793status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 794{ 795 // check calling permissions 796 if (!settingsAllowed()) { 797 return PERMISSION_DENIED; 798 } 799 800 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 801 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 802 ALOGE("setStreamMute() invalid stream %d", stream); 803 return BAD_VALUE; 804 } 805 806 AutoMutex lock(mLock); 807 mStreamTypes[stream].mute = muted; 808 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 809 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 810 811 return NO_ERROR; 812} 813 814float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return 0.0f; 818 } 819 820 AutoMutex lock(mLock); 821 float volume; 822 if (output) { 823 PlaybackThread *thread = checkPlaybackThread_l(output); 824 if (thread == NULL) { 825 return 0.0f; 826 } 827 volume = thread->streamVolume(stream); 828 } else { 829 volume = streamVolume_l(stream); 830 } 831 832 return volume; 833} 834 835bool AudioFlinger::streamMute(audio_stream_type_t stream) const 836{ 837 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 838 return true; 839 } 840 841 AutoMutex lock(mLock); 842 return streamMute_l(stream); 843} 844 845status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 846{ 847 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 848 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 849 // check calling permissions 850 if (!settingsAllowed()) { 851 return PERMISSION_DENIED; 852 } 853 854 // ioHandle == 0 means the parameters are global to the audio hardware interface 855 if (ioHandle == 0) { 856 Mutex::Autolock _l(mLock); 857 status_t final_result = NO_ERROR; 858 { 859 AutoMutex lock(mHardwareLock); 860 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 861 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 862 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 863 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 864 final_result = result ?: final_result; 865 } 866 mHardwareStatus = AUDIO_HW_IDLE; 867 } 868 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 869 AudioParameter param = AudioParameter(keyValuePairs); 870 String8 value; 871 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 872 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 873 if (mBtNrecIsOff != btNrecIsOff) { 874 for (size_t i = 0; i < mRecordThreads.size(); i++) { 875 sp<RecordThread> thread = mRecordThreads.valueAt(i); 876 RecordThread::RecordTrack *track = thread->track(); 877 if (track != NULL) { 878 audio_devices_t device = (audio_devices_t)( 879 thread->device() & AUDIO_DEVICE_IN_ALL); 880 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 881 thread->setEffectSuspended(FX_IID_AEC, 882 suspend, 883 track->sessionId()); 884 thread->setEffectSuspended(FX_IID_NS, 885 suspend, 886 track->sessionId()); 887 } 888 } 889 mBtNrecIsOff = btNrecIsOff; 890 } 891 } 892 return final_result; 893 } 894 895 // hold a strong ref on thread in case closeOutput() or closeInput() is called 896 // and the thread is exited once the lock is released 897 sp<ThreadBase> thread; 898 { 899 Mutex::Autolock _l(mLock); 900 thread = checkPlaybackThread_l(ioHandle); 901 if (thread == NULL) { 902 thread = checkRecordThread_l(ioHandle); 903 } else if (thread == primaryPlaybackThread_l()) { 904 // indicate output device change to all input threads for pre processing 905 AudioParameter param = AudioParameter(keyValuePairs); 906 int value; 907 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 908 (value != 0)) { 909 for (size_t i = 0; i < mRecordThreads.size(); i++) { 910 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 911 } 912 } 913 } 914 } 915 if (thread != 0) { 916 return thread->setParameters(keyValuePairs); 917 } 918 return BAD_VALUE; 919} 920 921String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 922{ 923// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 924// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 925 926 Mutex::Autolock _l(mLock); 927 928 if (ioHandle == 0) { 929 String8 out_s8; 930 931 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 932 char *s; 933 { 934 AutoMutex lock(mHardwareLock); 935 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 936 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 937 s = dev->get_parameters(dev, keys.string()); 938 mHardwareStatus = AUDIO_HW_IDLE; 939 } 940 out_s8 += String8(s ? s : ""); 941 free(s); 942 } 943 return out_s8; 944 } 945 946 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 947 if (playbackThread != NULL) { 948 return playbackThread->getParameters(keys); 949 } 950 RecordThread *recordThread = checkRecordThread_l(ioHandle); 951 if (recordThread != NULL) { 952 return recordThread->getParameters(keys); 953 } 954 return String8(""); 955} 956 957size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 958{ 959 status_t ret = initCheck(); 960 if (ret != NO_ERROR) { 961 return 0; 962 } 963 964 AutoMutex lock(mHardwareLock); 965 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 966 struct audio_config config = { 967 sample_rate: sampleRate, 968 channel_mask: audio_channel_in_mask_from_count(channelCount), 969 format: format, 970 }; 971 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 972 mHardwareStatus = AUDIO_HW_IDLE; 973 return size; 974} 975 976unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 977{ 978 if (ioHandle == 0) { 979 return 0; 980 } 981 982 Mutex::Autolock _l(mLock); 983 984 RecordThread *recordThread = checkRecordThread_l(ioHandle); 985 if (recordThread != NULL) { 986 return recordThread->getInputFramesLost(); 987 } 988 return 0; 989} 990 991status_t AudioFlinger::setVoiceVolume(float value) 992{ 993 status_t ret = initCheck(); 994 if (ret != NO_ERROR) { 995 return ret; 996 } 997 998 // check calling permissions 999 if (!settingsAllowed()) { 1000 return PERMISSION_DENIED; 1001 } 1002 1003 AutoMutex lock(mHardwareLock); 1004 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1005 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1006 mHardwareStatus = AUDIO_HW_IDLE; 1007 1008 return ret; 1009} 1010 1011status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1012 audio_io_handle_t output) const 1013{ 1014 status_t status; 1015 1016 Mutex::Autolock _l(mLock); 1017 1018 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1019 if (playbackThread != NULL) { 1020 return playbackThread->getRenderPosition(halFrames, dspFrames); 1021 } 1022 1023 return BAD_VALUE; 1024} 1025 1026void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1027{ 1028 1029 Mutex::Autolock _l(mLock); 1030 1031 pid_t pid = IPCThreadState::self()->getCallingPid(); 1032 if (mNotificationClients.indexOfKey(pid) < 0) { 1033 sp<NotificationClient> notificationClient = new NotificationClient(this, 1034 client, 1035 pid); 1036 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1037 1038 mNotificationClients.add(pid, notificationClient); 1039 1040 sp<IBinder> binder = client->asBinder(); 1041 binder->linkToDeath(notificationClient); 1042 1043 // the config change is always sent from playback or record threads to avoid deadlock 1044 // with AudioSystem::gLock 1045 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1046 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1047 } 1048 1049 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1050 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1051 } 1052 } 1053} 1054 1055void AudioFlinger::removeNotificationClient(pid_t pid) 1056{ 1057 Mutex::Autolock _l(mLock); 1058 1059 mNotificationClients.removeItem(pid); 1060 1061 ALOGV("%d died, releasing its sessions", pid); 1062 size_t num = mAudioSessionRefs.size(); 1063 bool removed = false; 1064 for (size_t i = 0; i< num; ) { 1065 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1066 ALOGV(" pid %d @ %d", ref->mPid, i); 1067 if (ref->mPid == pid) { 1068 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1069 mAudioSessionRefs.removeAt(i); 1070 delete ref; 1071 removed = true; 1072 num--; 1073 } else { 1074 i++; 1075 } 1076 } 1077 if (removed) { 1078 purgeStaleEffects_l(); 1079 } 1080} 1081 1082// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1083void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1084{ 1085 size_t size = mNotificationClients.size(); 1086 for (size_t i = 0; i < size; i++) { 1087 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1088 param2); 1089 } 1090} 1091 1092// removeClient_l() must be called with AudioFlinger::mLock held 1093void AudioFlinger::removeClient_l(pid_t pid) 1094{ 1095 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1096 mClients.removeItem(pid); 1097} 1098 1099 1100// ---------------------------------------------------------------------------- 1101 1102AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1103 uint32_t device, type_t type) 1104 : Thread(false), 1105 mType(type), 1106 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1107 // mChannelMask 1108 mChannelCount(0), 1109 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1110 mParamStatus(NO_ERROR), 1111 mStandby(false), mId(id), 1112 mDevice(device), 1113 mDeathRecipient(new PMDeathRecipient(this)) 1114{ 1115} 1116 1117AudioFlinger::ThreadBase::~ThreadBase() 1118{ 1119 mParamCond.broadcast(); 1120 // do not lock the mutex in destructor 1121 releaseWakeLock_l(); 1122 if (mPowerManager != 0) { 1123 sp<IBinder> binder = mPowerManager->asBinder(); 1124 binder->unlinkToDeath(mDeathRecipient); 1125 } 1126} 1127 1128void AudioFlinger::ThreadBase::exit() 1129{ 1130 ALOGV("ThreadBase::exit"); 1131 { 1132 // This lock prevents the following race in thread (uniprocessor for illustration): 1133 // if (!exitPending()) { 1134 // // context switch from here to exit() 1135 // // exit() calls requestExit(), what exitPending() observes 1136 // // exit() calls signal(), which is dropped since no waiters 1137 // // context switch back from exit() to here 1138 // mWaitWorkCV.wait(...); 1139 // // now thread is hung 1140 // } 1141 AutoMutex lock(mLock); 1142 requestExit(); 1143 mWaitWorkCV.signal(); 1144 } 1145 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1146 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1147 requestExitAndWait(); 1148} 1149 1150status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1151{ 1152 status_t status; 1153 1154 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1155 Mutex::Autolock _l(mLock); 1156 1157 mNewParameters.add(keyValuePairs); 1158 mWaitWorkCV.signal(); 1159 // wait condition with timeout in case the thread loop has exited 1160 // before the request could be processed 1161 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1162 status = mParamStatus; 1163 mWaitWorkCV.signal(); 1164 } else { 1165 status = TIMED_OUT; 1166 } 1167 return status; 1168} 1169 1170void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1171{ 1172 Mutex::Autolock _l(mLock); 1173 sendConfigEvent_l(event, param); 1174} 1175 1176// sendConfigEvent_l() must be called with ThreadBase::mLock held 1177void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1178{ 1179 ConfigEvent configEvent; 1180 configEvent.mEvent = event; 1181 configEvent.mParam = param; 1182 mConfigEvents.add(configEvent); 1183 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1184 mWaitWorkCV.signal(); 1185} 1186 1187void AudioFlinger::ThreadBase::processConfigEvents() 1188{ 1189 mLock.lock(); 1190 while (!mConfigEvents.isEmpty()) { 1191 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1192 ConfigEvent configEvent = mConfigEvents[0]; 1193 mConfigEvents.removeAt(0); 1194 // release mLock before locking AudioFlinger mLock: lock order is always 1195 // AudioFlinger then ThreadBase to avoid cross deadlock 1196 mLock.unlock(); 1197 mAudioFlinger->mLock.lock(); 1198 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1199 mAudioFlinger->mLock.unlock(); 1200 mLock.lock(); 1201 } 1202 mLock.unlock(); 1203} 1204 1205status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1206{ 1207 const size_t SIZE = 256; 1208 char buffer[SIZE]; 1209 String8 result; 1210 1211 bool locked = tryLock(mLock); 1212 if (!locked) { 1213 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1214 write(fd, buffer, strlen(buffer)); 1215 } 1216 1217 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1218 result.append(buffer); 1219 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1220 result.append(buffer); 1221 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1222 result.append(buffer); 1223 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1224 result.append(buffer); 1225 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1226 result.append(buffer); 1227 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1228 result.append(buffer); 1229 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1236 result.append(buffer); 1237 1238 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1239 result.append(buffer); 1240 result.append(" Index Command"); 1241 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1242 snprintf(buffer, SIZE, "\n %02d ", i); 1243 result.append(buffer); 1244 result.append(mNewParameters[i]); 1245 } 1246 1247 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1248 result.append(buffer); 1249 snprintf(buffer, SIZE, " Index event param\n"); 1250 result.append(buffer); 1251 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1252 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1253 result.append(buffer); 1254 } 1255 result.append("\n"); 1256 1257 write(fd, result.string(), result.size()); 1258 1259 if (locked) { 1260 mLock.unlock(); 1261 } 1262 return NO_ERROR; 1263} 1264 1265status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1266{ 1267 const size_t SIZE = 256; 1268 char buffer[SIZE]; 1269 String8 result; 1270 1271 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1272 write(fd, buffer, strlen(buffer)); 1273 1274 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1275 sp<EffectChain> chain = mEffectChains[i]; 1276 if (chain != 0) { 1277 chain->dump(fd, args); 1278 } 1279 } 1280 return NO_ERROR; 1281} 1282 1283void AudioFlinger::ThreadBase::acquireWakeLock() 1284{ 1285 Mutex::Autolock _l(mLock); 1286 acquireWakeLock_l(); 1287} 1288 1289void AudioFlinger::ThreadBase::acquireWakeLock_l() 1290{ 1291 if (mPowerManager == 0) { 1292 // use checkService() to avoid blocking if power service is not up yet 1293 sp<IBinder> binder = 1294 defaultServiceManager()->checkService(String16("power")); 1295 if (binder == 0) { 1296 ALOGW("Thread %s cannot connect to the power manager service", mName); 1297 } else { 1298 mPowerManager = interface_cast<IPowerManager>(binder); 1299 binder->linkToDeath(mDeathRecipient); 1300 } 1301 } 1302 if (mPowerManager != 0) { 1303 sp<IBinder> binder = new BBinder(); 1304 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1305 binder, 1306 String16(mName)); 1307 if (status == NO_ERROR) { 1308 mWakeLockToken = binder; 1309 } 1310 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1311 } 1312} 1313 1314void AudioFlinger::ThreadBase::releaseWakeLock() 1315{ 1316 Mutex::Autolock _l(mLock); 1317 releaseWakeLock_l(); 1318} 1319 1320void AudioFlinger::ThreadBase::releaseWakeLock_l() 1321{ 1322 if (mWakeLockToken != 0) { 1323 ALOGV("releaseWakeLock_l() %s", mName); 1324 if (mPowerManager != 0) { 1325 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1326 } 1327 mWakeLockToken.clear(); 1328 } 1329} 1330 1331void AudioFlinger::ThreadBase::clearPowerManager() 1332{ 1333 Mutex::Autolock _l(mLock); 1334 releaseWakeLock_l(); 1335 mPowerManager.clear(); 1336} 1337 1338void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1339{ 1340 sp<ThreadBase> thread = mThread.promote(); 1341 if (thread != 0) { 1342 thread->clearPowerManager(); 1343 } 1344 ALOGW("power manager service died !!!"); 1345} 1346 1347void AudioFlinger::ThreadBase::setEffectSuspended( 1348 const effect_uuid_t *type, bool suspend, int sessionId) 1349{ 1350 Mutex::Autolock _l(mLock); 1351 setEffectSuspended_l(type, suspend, sessionId); 1352} 1353 1354void AudioFlinger::ThreadBase::setEffectSuspended_l( 1355 const effect_uuid_t *type, bool suspend, int sessionId) 1356{ 1357 sp<EffectChain> chain = getEffectChain_l(sessionId); 1358 if (chain != 0) { 1359 if (type != NULL) { 1360 chain->setEffectSuspended_l(type, suspend); 1361 } else { 1362 chain->setEffectSuspendedAll_l(suspend); 1363 } 1364 } 1365 1366 updateSuspendedSessions_l(type, suspend, sessionId); 1367} 1368 1369void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1370{ 1371 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1372 if (index < 0) { 1373 return; 1374 } 1375 1376 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1377 mSuspendedSessions.editValueAt(index); 1378 1379 for (size_t i = 0; i < sessionEffects.size(); i++) { 1380 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1381 for (int j = 0; j < desc->mRefCount; j++) { 1382 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1383 chain->setEffectSuspendedAll_l(true); 1384 } else { 1385 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1386 desc->mType.timeLow); 1387 chain->setEffectSuspended_l(&desc->mType, true); 1388 } 1389 } 1390 } 1391} 1392 1393void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1394 bool suspend, 1395 int sessionId) 1396{ 1397 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1398 1399 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1400 1401 if (suspend) { 1402 if (index >= 0) { 1403 sessionEffects = mSuspendedSessions.editValueAt(index); 1404 } else { 1405 mSuspendedSessions.add(sessionId, sessionEffects); 1406 } 1407 } else { 1408 if (index < 0) { 1409 return; 1410 } 1411 sessionEffects = mSuspendedSessions.editValueAt(index); 1412 } 1413 1414 1415 int key = EffectChain::kKeyForSuspendAll; 1416 if (type != NULL) { 1417 key = type->timeLow; 1418 } 1419 index = sessionEffects.indexOfKey(key); 1420 1421 sp<SuspendedSessionDesc> desc; 1422 if (suspend) { 1423 if (index >= 0) { 1424 desc = sessionEffects.valueAt(index); 1425 } else { 1426 desc = new SuspendedSessionDesc(); 1427 if (type != NULL) { 1428 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1429 } 1430 sessionEffects.add(key, desc); 1431 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1432 } 1433 desc->mRefCount++; 1434 } else { 1435 if (index < 0) { 1436 return; 1437 } 1438 desc = sessionEffects.valueAt(index); 1439 if (--desc->mRefCount == 0) { 1440 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1441 sessionEffects.removeItemsAt(index); 1442 if (sessionEffects.isEmpty()) { 1443 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1444 sessionId); 1445 mSuspendedSessions.removeItem(sessionId); 1446 } 1447 } 1448 } 1449 if (!sessionEffects.isEmpty()) { 1450 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1451 } 1452} 1453 1454void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1455 bool enabled, 1456 int sessionId) 1457{ 1458 Mutex::Autolock _l(mLock); 1459 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1460} 1461 1462void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1463 bool enabled, 1464 int sessionId) 1465{ 1466 if (mType != RECORD) { 1467 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1468 // another session. This gives the priority to well behaved effect control panels 1469 // and applications not using global effects. 1470 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1471 // global effects 1472 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1473 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1474 } 1475 } 1476 1477 sp<EffectChain> chain = getEffectChain_l(sessionId); 1478 if (chain != 0) { 1479 chain->checkSuspendOnEffectEnabled(effect, enabled); 1480 } 1481} 1482 1483// ---------------------------------------------------------------------------- 1484 1485AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1486 AudioStreamOut* output, 1487 audio_io_handle_t id, 1488 uint32_t device, 1489 type_t type) 1490 : ThreadBase(audioFlinger, id, device, type), 1491 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1492 // Assumes constructor is called by AudioFlinger with it's mLock held, 1493 // but it would be safer to explicitly pass initial masterMute as parameter 1494 mMasterMute(audioFlinger->masterMute_l()), 1495 // mStreamTypes[] initialized in constructor body 1496 mOutput(output), 1497 // Assumes constructor is called by AudioFlinger with it's mLock held, 1498 // but it would be safer to explicitly pass initial masterVolume as parameter 1499 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1500 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1501 mMixerStatus(MIXER_IDLE), 1502 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1503 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1504 // index 0 is reserved for normal mixer's submix 1505 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1506{ 1507 snprintf(mName, kNameLength, "AudioOut_%X", id); 1508 1509 readOutputParameters(); 1510 1511 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1512 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1513 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1514 stream = (audio_stream_type_t) (stream + 1)) { 1515 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1516 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1517 } 1518 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1519 // because mAudioFlinger doesn't have one to copy from 1520} 1521 1522AudioFlinger::PlaybackThread::~PlaybackThread() 1523{ 1524 delete [] mMixBuffer; 1525} 1526 1527status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1528{ 1529 dumpInternals(fd, args); 1530 dumpTracks(fd, args); 1531 dumpEffectChains(fd, args); 1532 return NO_ERROR; 1533} 1534 1535status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1536{ 1537 const size_t SIZE = 256; 1538 char buffer[SIZE]; 1539 String8 result; 1540 1541 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1542 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1543 const stream_type_t *st = &mStreamTypes[i]; 1544 if (i > 0) { 1545 result.appendFormat(", "); 1546 } 1547 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1548 if (st->mute) { 1549 result.append("M"); 1550 } 1551 } 1552 result.append("\n"); 1553 write(fd, result.string(), result.length()); 1554 result.clear(); 1555 1556 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1557 result.append(buffer); 1558 Track::appendDumpHeader(result); 1559 for (size_t i = 0; i < mTracks.size(); ++i) { 1560 sp<Track> track = mTracks[i]; 1561 if (track != 0) { 1562 track->dump(buffer, SIZE); 1563 result.append(buffer); 1564 } 1565 } 1566 1567 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1568 result.append(buffer); 1569 Track::appendDumpHeader(result); 1570 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1571 sp<Track> track = mActiveTracks[i].promote(); 1572 if (track != 0) { 1573 track->dump(buffer, SIZE); 1574 result.append(buffer); 1575 } 1576 } 1577 write(fd, result.string(), result.size()); 1578 1579 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1580 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1581 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1582 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1583 1584 return NO_ERROR; 1585} 1586 1587status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1588{ 1589 const size_t SIZE = 256; 1590 char buffer[SIZE]; 1591 String8 result; 1592 1593 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1594 result.append(buffer); 1595 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1596 result.append(buffer); 1597 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1598 result.append(buffer); 1599 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1600 result.append(buffer); 1601 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1602 result.append(buffer); 1603 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1604 result.append(buffer); 1605 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1606 result.append(buffer); 1607 write(fd, result.string(), result.size()); 1608 1609 dumpBase(fd, args); 1610 1611 return NO_ERROR; 1612} 1613 1614// Thread virtuals 1615status_t AudioFlinger::PlaybackThread::readyToRun() 1616{ 1617 status_t status = initCheck(); 1618 if (status == NO_ERROR) { 1619 ALOGI("AudioFlinger's thread %p ready to run", this); 1620 } else { 1621 ALOGE("No working audio driver found."); 1622 } 1623 return status; 1624} 1625 1626void AudioFlinger::PlaybackThread::onFirstRef() 1627{ 1628 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1629} 1630 1631// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1632sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1633 const sp<AudioFlinger::Client>& client, 1634 audio_stream_type_t streamType, 1635 uint32_t sampleRate, 1636 audio_format_t format, 1637 uint32_t channelMask, 1638 int frameCount, 1639 const sp<IMemory>& sharedBuffer, 1640 int sessionId, 1641 IAudioFlinger::track_flags_t flags, 1642 pid_t tid, 1643 status_t *status) 1644{ 1645 sp<Track> track; 1646 status_t lStatus; 1647 1648 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1649 1650 // client expresses a preference for FAST, but we get the final say 1651 if (flags & IAudioFlinger::TRACK_FAST) { 1652 if ( 1653 // not timed 1654 (!isTimed) && 1655 // either of these use cases: 1656 ( 1657 // use case 1: shared buffer with any frame count 1658 ( 1659 (sharedBuffer != 0) 1660 ) || 1661 // use case 2: callback handler and frame count is default or at least as large as HAL 1662 ( 1663 (tid != -1) && 1664 ((frameCount == 0) || 1665 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1666 ) 1667 ) && 1668 // PCM data 1669 audio_is_linear_pcm(format) && 1670 // mono or stereo 1671 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1672 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1673#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1674 // hardware sample rate 1675 (sampleRate == mSampleRate) && 1676#endif 1677 // normal mixer has an associated fast mixer 1678 hasFastMixer() && 1679 // there are sufficient fast track slots available 1680 (mFastTrackAvailMask != 0) 1681 // FIXME test that MixerThread for this fast track has a capable output HAL 1682 // FIXME add a permission test also? 1683 ) { 1684 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1685 if (frameCount == 0) { 1686 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1687 } 1688 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1689 frameCount, mFrameCount); 1690 } else { 1691 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1692 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1693 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1694 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1695 audio_is_linear_pcm(format), 1696 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1697 flags &= ~IAudioFlinger::TRACK_FAST; 1698 // For compatibility with AudioTrack calculation, buffer depth is forced 1699 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1700 // This is probably too conservative, but legacy application code may depend on it. 1701 // If you change this calculation, also review the start threshold which is related. 1702 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1703 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1704 if (minBufCount < 2) { 1705 minBufCount = 2; 1706 } 1707 int minFrameCount = mNormalFrameCount * minBufCount; 1708 if (frameCount < minFrameCount) { 1709 frameCount = minFrameCount; 1710 } 1711 } 1712 } 1713 1714 if (mType == DIRECT) { 1715 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1716 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1717 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1718 "for output %p with format %d", 1719 sampleRate, format, channelMask, mOutput, mFormat); 1720 lStatus = BAD_VALUE; 1721 goto Exit; 1722 } 1723 } 1724 } else { 1725 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1726 if (sampleRate > mSampleRate*2) { 1727 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1728 lStatus = BAD_VALUE; 1729 goto Exit; 1730 } 1731 } 1732 1733 lStatus = initCheck(); 1734 if (lStatus != NO_ERROR) { 1735 ALOGE("Audio driver not initialized."); 1736 goto Exit; 1737 } 1738 1739 { // scope for mLock 1740 Mutex::Autolock _l(mLock); 1741 1742 // all tracks in same audio session must share the same routing strategy otherwise 1743 // conflicts will happen when tracks are moved from one output to another by audio policy 1744 // manager 1745 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1746 for (size_t i = 0; i < mTracks.size(); ++i) { 1747 sp<Track> t = mTracks[i]; 1748 if (t != 0 && !t->isOutputTrack()) { 1749 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1750 if (sessionId == t->sessionId() && strategy != actual) { 1751 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1752 strategy, actual); 1753 lStatus = BAD_VALUE; 1754 goto Exit; 1755 } 1756 } 1757 } 1758 1759 if (!isTimed) { 1760 track = new Track(this, client, streamType, sampleRate, format, 1761 channelMask, frameCount, sharedBuffer, sessionId, flags); 1762 } else { 1763 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1764 channelMask, frameCount, sharedBuffer, sessionId); 1765 } 1766 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1767 lStatus = NO_MEMORY; 1768 goto Exit; 1769 } 1770 mTracks.add(track); 1771 1772 sp<EffectChain> chain = getEffectChain_l(sessionId); 1773 if (chain != 0) { 1774 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1775 track->setMainBuffer(chain->inBuffer()); 1776 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1777 chain->incTrackCnt(); 1778 } 1779 } 1780 1781#ifdef HAVE_REQUEST_PRIORITY 1782 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1783 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1784 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1785 // so ask activity manager to do this on our behalf 1786 int err = requestPriority(callingPid, tid, 1); 1787 if (err != 0) { 1788 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1789 1, callingPid, tid, err); 1790 } 1791 } 1792#endif 1793 1794 lStatus = NO_ERROR; 1795 1796Exit: 1797 if (status) { 1798 *status = lStatus; 1799 } 1800 return track; 1801} 1802 1803uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1804{ 1805 if (mFastMixer != NULL) { 1806 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1807 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1808 } 1809 return latency; 1810} 1811 1812uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1813{ 1814 return latency; 1815} 1816 1817uint32_t AudioFlinger::PlaybackThread::latency() const 1818{ 1819 Mutex::Autolock _l(mLock); 1820 if (initCheck() == NO_ERROR) { 1821 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1822 } else { 1823 return 0; 1824 } 1825} 1826 1827void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1828{ 1829 Mutex::Autolock _l(mLock); 1830 mMasterVolume = value; 1831} 1832 1833void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1834{ 1835 Mutex::Autolock _l(mLock); 1836 setMasterMute_l(muted); 1837} 1838 1839void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1840{ 1841 Mutex::Autolock _l(mLock); 1842 mStreamTypes[stream].volume = value; 1843} 1844 1845void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1846{ 1847 Mutex::Autolock _l(mLock); 1848 mStreamTypes[stream].mute = muted; 1849} 1850 1851float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1852{ 1853 Mutex::Autolock _l(mLock); 1854 return mStreamTypes[stream].volume; 1855} 1856 1857// addTrack_l() must be called with ThreadBase::mLock held 1858status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1859{ 1860 status_t status = ALREADY_EXISTS; 1861 1862 // set retry count for buffer fill 1863 track->mRetryCount = kMaxTrackStartupRetries; 1864 if (mActiveTracks.indexOf(track) < 0) { 1865 // the track is newly added, make sure it fills up all its 1866 // buffers before playing. This is to ensure the client will 1867 // effectively get the latency it requested. 1868 track->mFillingUpStatus = Track::FS_FILLING; 1869 track->mResetDone = false; 1870 track->mPresentationCompleteFrames = 0; 1871 mActiveTracks.add(track); 1872 if (track->mainBuffer() != mMixBuffer) { 1873 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1874 if (chain != 0) { 1875 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1876 chain->incActiveTrackCnt(); 1877 } 1878 } 1879 1880 status = NO_ERROR; 1881 } 1882 1883 ALOGV("mWaitWorkCV.broadcast"); 1884 mWaitWorkCV.broadcast(); 1885 1886 return status; 1887} 1888 1889// destroyTrack_l() must be called with ThreadBase::mLock held 1890void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1891{ 1892 track->mState = TrackBase::TERMINATED; 1893 // active tracks are removed by threadLoop() 1894 if (mActiveTracks.indexOf(track) < 0) { 1895 removeTrack_l(track); 1896 } 1897} 1898 1899void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1900{ 1901 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1902 mTracks.remove(track); 1903 deleteTrackName_l(track->name()); 1904 // redundant as track is about to be destroyed, for dumpsys only 1905 track->mName = -1; 1906 if (track->isFastTrack()) { 1907 int index = track->mFastIndex; 1908 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1909 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1910 mFastTrackAvailMask |= 1 << index; 1911 // redundant as track is about to be destroyed, for dumpsys only 1912 track->mFastIndex = -1; 1913 } 1914 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1915 if (chain != 0) { 1916 chain->decTrackCnt(); 1917 } 1918} 1919 1920String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1921{ 1922 String8 out_s8 = String8(""); 1923 char *s; 1924 1925 Mutex::Autolock _l(mLock); 1926 if (initCheck() != NO_ERROR) { 1927 return out_s8; 1928 } 1929 1930 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1931 out_s8 = String8(s); 1932 free(s); 1933 return out_s8; 1934} 1935 1936// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1937void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1938 AudioSystem::OutputDescriptor desc; 1939 void *param2 = NULL; 1940 1941 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1942 1943 switch (event) { 1944 case AudioSystem::OUTPUT_OPENED: 1945 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1946 desc.channels = mChannelMask; 1947 desc.samplingRate = mSampleRate; 1948 desc.format = mFormat; 1949 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1950 desc.latency = latency(); 1951 param2 = &desc; 1952 break; 1953 1954 case AudioSystem::STREAM_CONFIG_CHANGED: 1955 param2 = ¶m; 1956 case AudioSystem::OUTPUT_CLOSED: 1957 default: 1958 break; 1959 } 1960 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1961} 1962 1963void AudioFlinger::PlaybackThread::readOutputParameters() 1964{ 1965 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1966 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1967 mChannelCount = (uint16_t)popcount(mChannelMask); 1968 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1969 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1970 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1971 if (mFrameCount & 15) { 1972 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1973 mFrameCount); 1974 } 1975 1976 // Calculate size of normal mix buffer relative to the HAL output buffer size 1977 double multiplier = 1.0; 1978 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1979 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1980 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1981 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1982 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1983 maxNormalFrameCount = maxNormalFrameCount & ~15; 1984 if (maxNormalFrameCount < minNormalFrameCount) { 1985 maxNormalFrameCount = minNormalFrameCount; 1986 } 1987 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1988 if (multiplier <= 1.0) { 1989 multiplier = 1.0; 1990 } else if (multiplier <= 2.0) { 1991 if (2 * mFrameCount <= maxNormalFrameCount) { 1992 multiplier = 2.0; 1993 } else { 1994 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1995 } 1996 } else { 1997 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 1998 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 1999 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2000 // FIXME this rounding up should not be done if no HAL SRC 2001 uint32_t truncMult = (uint32_t) multiplier; 2002 if ((truncMult & 1)) { 2003 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2004 ++truncMult; 2005 } 2006 } 2007 multiplier = (double) truncMult; 2008 } 2009 } 2010 mNormalFrameCount = multiplier * mFrameCount; 2011 // round up to nearest 16 frames to satisfy AudioMixer 2012 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2013 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2014 2015 // FIXME - Current mixer implementation only supports stereo output: Always 2016 // Allocate a stereo buffer even if HW output is mono. 2017 delete[] mMixBuffer; 2018 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 2019 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 2020 2021 // force reconfiguration of effect chains and engines to take new buffer size and audio 2022 // parameters into account 2023 // Note that mLock is not held when readOutputParameters() is called from the constructor 2024 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2025 // matter. 2026 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2027 Vector< sp<EffectChain> > effectChains = mEffectChains; 2028 for (size_t i = 0; i < effectChains.size(); i ++) { 2029 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2030 } 2031} 2032 2033 2034status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2035{ 2036 if (halFrames == NULL || dspFrames == NULL) { 2037 return BAD_VALUE; 2038 } 2039 Mutex::Autolock _l(mLock); 2040 if (initCheck() != NO_ERROR) { 2041 return INVALID_OPERATION; 2042 } 2043 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2044 2045 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2046} 2047 2048uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2049{ 2050 Mutex::Autolock _l(mLock); 2051 uint32_t result = 0; 2052 if (getEffectChain_l(sessionId) != 0) { 2053 result = EFFECT_SESSION; 2054 } 2055 2056 for (size_t i = 0; i < mTracks.size(); ++i) { 2057 sp<Track> track = mTracks[i]; 2058 if (sessionId == track->sessionId() && 2059 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2060 result |= TRACK_SESSION; 2061 break; 2062 } 2063 } 2064 2065 return result; 2066} 2067 2068uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2069{ 2070 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2071 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2072 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2073 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2074 } 2075 for (size_t i = 0; i < mTracks.size(); i++) { 2076 sp<Track> track = mTracks[i]; 2077 if (sessionId == track->sessionId() && 2078 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2079 return AudioSystem::getStrategyForStream(track->streamType()); 2080 } 2081 } 2082 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2083} 2084 2085 2086AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2087{ 2088 Mutex::Autolock _l(mLock); 2089 return mOutput; 2090} 2091 2092AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2093{ 2094 Mutex::Autolock _l(mLock); 2095 AudioStreamOut *output = mOutput; 2096 mOutput = NULL; 2097 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2098 // must push a NULL and wait for ack 2099 mOutputSink.clear(); 2100 mPipeSink.clear(); 2101 mNormalSink.clear(); 2102 return output; 2103} 2104 2105// this method must always be called either with ThreadBase mLock held or inside the thread loop 2106audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2107{ 2108 if (mOutput == NULL) { 2109 return NULL; 2110 } 2111 return &mOutput->stream->common; 2112} 2113 2114uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2115{ 2116 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2117 // decoding and transfer time. So sleeping for half of the latency would likely cause 2118 // underruns 2119 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2120 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2121 } else { 2122 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2123 } 2124} 2125 2126status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2127{ 2128 if (!isValidSyncEvent(event)) { 2129 return BAD_VALUE; 2130 } 2131 2132 Mutex::Autolock _l(mLock); 2133 2134 for (size_t i = 0; i < mTracks.size(); ++i) { 2135 sp<Track> track = mTracks[i]; 2136 if (event->triggerSession() == track->sessionId()) { 2137 track->setSyncEvent(event); 2138 return NO_ERROR; 2139 } 2140 } 2141 2142 return NAME_NOT_FOUND; 2143} 2144 2145bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2146{ 2147 switch (event->type()) { 2148 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2149 return true; 2150 default: 2151 break; 2152 } 2153 return false; 2154} 2155 2156void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2157{ 2158 size_t count = tracksToRemove.size(); 2159 if (CC_UNLIKELY(count)) { 2160 for (size_t i = 0 ; i < count ; i++) { 2161 const sp<Track>& track = tracksToRemove.itemAt(i); 2162 if ((track->sharedBuffer() != 0) && 2163 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2164 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2165 } 2166 } 2167 } 2168 2169} 2170 2171// ---------------------------------------------------------------------------- 2172 2173AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2174 audio_io_handle_t id, uint32_t device, type_t type) 2175 : PlaybackThread(audioFlinger, output, id, device, type), 2176 // mAudioMixer below 2177#ifdef SOAKER 2178 mSoaker(NULL), 2179#endif 2180 // mFastMixer below 2181 mFastMixerFutex(0) 2182 // mOutputSink below 2183 // mPipeSink below 2184 // mNormalSink below 2185{ 2186 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2187 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2188 "mFrameCount=%d, mNormalFrameCount=%d", 2189 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2190 mNormalFrameCount); 2191 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2192 2193 // FIXME - Current mixer implementation only supports stereo output 2194 if (mChannelCount == 1) { 2195 ALOGE("Invalid audio hardware channel count"); 2196 } 2197 2198 // create an NBAIO sink for the HAL output stream, and negotiate 2199 mOutputSink = new AudioStreamOutSink(output->stream); 2200 size_t numCounterOffers = 0; 2201 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2202 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2203 ALOG_ASSERT(index == 0); 2204 2205 // initialize fast mixer depending on configuration 2206 bool initFastMixer; 2207 switch (kUseFastMixer) { 2208 case FastMixer_Never: 2209 initFastMixer = false; 2210 break; 2211 case FastMixer_Always: 2212 initFastMixer = true; 2213 break; 2214 case FastMixer_Static: 2215 case FastMixer_Dynamic: 2216 initFastMixer = mFrameCount < mNormalFrameCount; 2217 break; 2218 } 2219 if (initFastMixer) { 2220 2221 // create a MonoPipe to connect our submix to FastMixer 2222 NBAIO_Format format = mOutputSink->format(); 2223 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2224 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2225 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2226 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2227 const NBAIO_Format offers[1] = {format}; 2228 size_t numCounterOffers = 0; 2229 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2230 ALOG_ASSERT(index == 0); 2231 mPipeSink = monoPipe; 2232 2233#ifdef TEE_SINK_FRAMES 2234 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2235 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2236 numCounterOffers = 0; 2237 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2238 ALOG_ASSERT(index == 0); 2239 mTeeSink = teeSink; 2240 PipeReader *teeSource = new PipeReader(*teeSink); 2241 numCounterOffers = 0; 2242 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2243 ALOG_ASSERT(index == 0); 2244 mTeeSource = teeSource; 2245#endif 2246 2247#ifdef SOAKER 2248 // create a soaker as workaround for governor issues 2249 mSoaker = new Soaker(); 2250 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2251 mSoaker->run("Soaker", PRIORITY_LOWEST); 2252#endif 2253 2254 // create fast mixer and configure it initially with just one fast track for our submix 2255 mFastMixer = new FastMixer(); 2256 FastMixerStateQueue *sq = mFastMixer->sq(); 2257 FastMixerState *state = sq->begin(); 2258 FastTrack *fastTrack = &state->mFastTracks[0]; 2259 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2260 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2261 fastTrack->mVolumeProvider = NULL; 2262 fastTrack->mGeneration++; 2263 state->mFastTracksGen++; 2264 state->mTrackMask = 1; 2265 // fast mixer will use the HAL output sink 2266 state->mOutputSink = mOutputSink.get(); 2267 state->mOutputSinkGen++; 2268 state->mFrameCount = mFrameCount; 2269 state->mCommand = FastMixerState::COLD_IDLE; 2270 // already done in constructor initialization list 2271 //mFastMixerFutex = 0; 2272 state->mColdFutexAddr = &mFastMixerFutex; 2273 state->mColdGen++; 2274 state->mDumpState = &mFastMixerDumpState; 2275 state->mTeeSink = mTeeSink.get(); 2276 sq->end(); 2277 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2278 2279 // start the fast mixer 2280 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2281#ifdef HAVE_REQUEST_PRIORITY 2282 pid_t tid = mFastMixer->getTid(); 2283 int err = requestPriority(getpid_cached, tid, 2); 2284 if (err != 0) { 2285 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2286 2, getpid_cached, tid, err); 2287 } 2288#endif 2289 2290 } else { 2291 mFastMixer = NULL; 2292 } 2293 2294 switch (kUseFastMixer) { 2295 case FastMixer_Never: 2296 case FastMixer_Dynamic: 2297 mNormalSink = mOutputSink; 2298 break; 2299 case FastMixer_Always: 2300 mNormalSink = mPipeSink; 2301 break; 2302 case FastMixer_Static: 2303 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2304 break; 2305 } 2306} 2307 2308AudioFlinger::MixerThread::~MixerThread() 2309{ 2310 if (mFastMixer != NULL) { 2311 FastMixerStateQueue *sq = mFastMixer->sq(); 2312 FastMixerState *state = sq->begin(); 2313 if (state->mCommand == FastMixerState::COLD_IDLE) { 2314 int32_t old = android_atomic_inc(&mFastMixerFutex); 2315 if (old == -1) { 2316 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2317 } 2318 } 2319 state->mCommand = FastMixerState::EXIT; 2320 sq->end(); 2321 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2322 mFastMixer->join(); 2323 // Though the fast mixer thread has exited, it's state queue is still valid. 2324 // We'll use that extract the final state which contains one remaining fast track 2325 // corresponding to our sub-mix. 2326 state = sq->begin(); 2327 ALOG_ASSERT(state->mTrackMask == 1); 2328 FastTrack *fastTrack = &state->mFastTracks[0]; 2329 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2330 delete fastTrack->mBufferProvider; 2331 sq->end(false /*didModify*/); 2332 delete mFastMixer; 2333#ifdef SOAKER 2334 if (mSoaker != NULL) { 2335 mSoaker->requestExitAndWait(); 2336 } 2337 delete mSoaker; 2338#endif 2339 } 2340 delete mAudioMixer; 2341} 2342 2343class CpuStats { 2344public: 2345 CpuStats(); 2346 void sample(const String8 &title); 2347#ifdef DEBUG_CPU_USAGE 2348private: 2349 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2350 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2351 2352 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2353 2354 int mCpuNum; // thread's current CPU number 2355 int mCpukHz; // frequency of thread's current CPU in kHz 2356#endif 2357}; 2358 2359CpuStats::CpuStats() 2360#ifdef DEBUG_CPU_USAGE 2361 : mCpuNum(-1), mCpukHz(-1) 2362#endif 2363{ 2364} 2365 2366void CpuStats::sample(const String8 &title) { 2367#ifdef DEBUG_CPU_USAGE 2368 // get current thread's delta CPU time in wall clock ns 2369 double wcNs; 2370 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2371 2372 // record sample for wall clock statistics 2373 if (valid) { 2374 mWcStats.sample(wcNs); 2375 } 2376 2377 // get the current CPU number 2378 int cpuNum = sched_getcpu(); 2379 2380 // get the current CPU frequency in kHz 2381 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2382 2383 // check if either CPU number or frequency changed 2384 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2385 mCpuNum = cpuNum; 2386 mCpukHz = cpukHz; 2387 // ignore sample for purposes of cycles 2388 valid = false; 2389 } 2390 2391 // if no change in CPU number or frequency, then record sample for cycle statistics 2392 if (valid && mCpukHz > 0) { 2393 double cycles = wcNs * cpukHz * 0.000001; 2394 mHzStats.sample(cycles); 2395 } 2396 2397 unsigned n = mWcStats.n(); 2398 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2399 if ((n & 127) == 1) { 2400 long long elapsed = mCpuUsage.elapsed(); 2401 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2402 double perLoop = elapsed / (double) n; 2403 double perLoop100 = perLoop * 0.01; 2404 double perLoop1k = perLoop * 0.001; 2405 double mean = mWcStats.mean(); 2406 double stddev = mWcStats.stddev(); 2407 double minimum = mWcStats.minimum(); 2408 double maximum = mWcStats.maximum(); 2409 double meanCycles = mHzStats.mean(); 2410 double stddevCycles = mHzStats.stddev(); 2411 double minCycles = mHzStats.minimum(); 2412 double maxCycles = mHzStats.maximum(); 2413 mCpuUsage.resetElapsed(); 2414 mWcStats.reset(); 2415 mHzStats.reset(); 2416 ALOGD("CPU usage for %s over past %.1f secs\n" 2417 " (%u mixer loops at %.1f mean ms per loop):\n" 2418 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2419 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2420 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2421 title.string(), 2422 elapsed * .000000001, n, perLoop * .000001, 2423 mean * .001, 2424 stddev * .001, 2425 minimum * .001, 2426 maximum * .001, 2427 mean / perLoop100, 2428 stddev / perLoop100, 2429 minimum / perLoop100, 2430 maximum / perLoop100, 2431 meanCycles / perLoop1k, 2432 stddevCycles / perLoop1k, 2433 minCycles / perLoop1k, 2434 maxCycles / perLoop1k); 2435 2436 } 2437 } 2438#endif 2439}; 2440 2441void AudioFlinger::PlaybackThread::checkSilentMode_l() 2442{ 2443 if (!mMasterMute) { 2444 char value[PROPERTY_VALUE_MAX]; 2445 if (property_get("ro.audio.silent", value, "0") > 0) { 2446 char *endptr; 2447 unsigned long ul = strtoul(value, &endptr, 0); 2448 if (*endptr == '\0' && ul != 0) { 2449 ALOGD("Silence is golden"); 2450 // The setprop command will not allow a property to be changed after 2451 // the first time it is set, so we don't have to worry about un-muting. 2452 setMasterMute_l(true); 2453 } 2454 } 2455 } 2456} 2457 2458bool AudioFlinger::PlaybackThread::threadLoop() 2459{ 2460 Vector< sp<Track> > tracksToRemove; 2461 2462 standbyTime = systemTime(); 2463 2464 // MIXER 2465 nsecs_t lastWarning = 0; 2466if (mType == MIXER) { 2467 longStandbyExit = false; 2468} 2469 2470 // DUPLICATING 2471 // FIXME could this be made local to while loop? 2472 writeFrames = 0; 2473 2474 cacheParameters_l(); 2475 sleepTime = idleSleepTime; 2476 2477if (mType == MIXER) { 2478 sleepTimeShift = 0; 2479} 2480 2481 CpuStats cpuStats; 2482 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2483 2484 acquireWakeLock(); 2485 2486 while (!exitPending()) 2487 { 2488 cpuStats.sample(myName); 2489 2490 Vector< sp<EffectChain> > effectChains; 2491 2492 processConfigEvents(); 2493 2494 { // scope for mLock 2495 2496 Mutex::Autolock _l(mLock); 2497 2498 if (checkForNewParameters_l()) { 2499 cacheParameters_l(); 2500 } 2501 2502 saveOutputTracks(); 2503 2504 // put audio hardware into standby after short delay 2505 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2506 mSuspended > 0)) { 2507 if (!mStandby) { 2508 2509 threadLoop_standby(); 2510 2511 mStandby = true; 2512 mBytesWritten = 0; 2513 } 2514 2515 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2516 // we're about to wait, flush the binder command buffer 2517 IPCThreadState::self()->flushCommands(); 2518 2519 clearOutputTracks(); 2520 2521 if (exitPending()) break; 2522 2523 releaseWakeLock_l(); 2524 // wait until we have something to do... 2525 ALOGV("%s going to sleep", myName.string()); 2526 mWaitWorkCV.wait(mLock); 2527 ALOGV("%s waking up", myName.string()); 2528 acquireWakeLock_l(); 2529 2530 mMixerStatus = MIXER_IDLE; 2531 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2532 2533 checkSilentMode_l(); 2534 2535 standbyTime = systemTime() + standbyDelay; 2536 sleepTime = idleSleepTime; 2537 if (mType == MIXER) { 2538 sleepTimeShift = 0; 2539 } 2540 2541 continue; 2542 } 2543 } 2544 2545 // mMixerStatusIgnoringFastTracks is also updated internally 2546 mMixerStatus = prepareTracks_l(&tracksToRemove); 2547 2548 // prevent any changes in effect chain list and in each effect chain 2549 // during mixing and effect process as the audio buffers could be deleted 2550 // or modified if an effect is created or deleted 2551 lockEffectChains_l(effectChains); 2552 } 2553 2554 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2555 threadLoop_mix(); 2556 } else { 2557 threadLoop_sleepTime(); 2558 } 2559 2560 if (mSuspended > 0) { 2561 sleepTime = suspendSleepTimeUs(); 2562 } 2563 2564 // only process effects if we're going to write 2565 if (sleepTime == 0) { 2566 for (size_t i = 0; i < effectChains.size(); i ++) { 2567 effectChains[i]->process_l(); 2568 } 2569 } 2570 2571 // enable changes in effect chain 2572 unlockEffectChains(effectChains); 2573 2574 // sleepTime == 0 means we must write to audio hardware 2575 if (sleepTime == 0) { 2576 2577 threadLoop_write(); 2578 2579if (mType == MIXER) { 2580 // write blocked detection 2581 nsecs_t now = systemTime(); 2582 nsecs_t delta = now - mLastWriteTime; 2583 if (!mStandby && delta > maxPeriod) { 2584 mNumDelayedWrites++; 2585 if ((now - lastWarning) > kWarningThrottleNs) { 2586#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2587 ScopedTrace st(ATRACE_TAG, "underrun"); 2588#endif 2589 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2590 ns2ms(delta), mNumDelayedWrites, this); 2591 lastWarning = now; 2592 } 2593 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2594 // a different threshold. Or completely removed for what it is worth anyway... 2595 if (mStandby) { 2596 longStandbyExit = true; 2597 } 2598 } 2599} 2600 2601 mStandby = false; 2602 } else { 2603 usleep(sleepTime); 2604 } 2605 2606 // Finally let go of removed track(s), without the lock held 2607 // since we can't guarantee the destructors won't acquire that 2608 // same lock. This will also mutate and push a new fast mixer state. 2609 threadLoop_removeTracks(tracksToRemove); 2610 tracksToRemove.clear(); 2611 2612 // FIXME I don't understand the need for this here; 2613 // it was in the original code but maybe the 2614 // assignment in saveOutputTracks() makes this unnecessary? 2615 clearOutputTracks(); 2616 2617 // Effect chains will be actually deleted here if they were removed from 2618 // mEffectChains list during mixing or effects processing 2619 effectChains.clear(); 2620 2621 // FIXME Note that the above .clear() is no longer necessary since effectChains 2622 // is now local to this block, but will keep it for now (at least until merge done). 2623 } 2624 2625if (mType == MIXER || mType == DIRECT) { 2626 // put output stream into standby mode 2627 if (!mStandby) { 2628 mOutput->stream->common.standby(&mOutput->stream->common); 2629 } 2630} 2631if (mType == DUPLICATING) { 2632 // for DuplicatingThread, standby mode is handled by the outputTracks 2633} 2634 2635 releaseWakeLock(); 2636 2637 ALOGV("Thread %p type %d exiting", this, mType); 2638 return false; 2639} 2640 2641void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2642{ 2643 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2644} 2645 2646void AudioFlinger::MixerThread::threadLoop_write() 2647{ 2648 // FIXME we should only do one push per cycle; confirm this is true 2649 // Start the fast mixer if it's not already running 2650 if (mFastMixer != NULL) { 2651 FastMixerStateQueue *sq = mFastMixer->sq(); 2652 FastMixerState *state = sq->begin(); 2653 if (state->mCommand != FastMixerState::MIX_WRITE && 2654 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2655 if (state->mCommand == FastMixerState::COLD_IDLE) { 2656 int32_t old = android_atomic_inc(&mFastMixerFutex); 2657 if (old == -1) { 2658 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2659 } 2660 } 2661 state->mCommand = FastMixerState::MIX_WRITE; 2662 sq->end(); 2663 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2664 if (kUseFastMixer == FastMixer_Dynamic) { 2665 mNormalSink = mPipeSink; 2666 } 2667 } else { 2668 sq->end(false /*didModify*/); 2669 } 2670 } 2671 PlaybackThread::threadLoop_write(); 2672} 2673 2674// shared by MIXER and DIRECT, overridden by DUPLICATING 2675void AudioFlinger::PlaybackThread::threadLoop_write() 2676{ 2677 // FIXME rewrite to reduce number of system calls 2678 mLastWriteTime = systemTime(); 2679 mInWrite = true; 2680 2681#define mBitShift 2 // FIXME 2682 size_t count = mixBufferSize >> mBitShift; 2683#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2684 Tracer::traceBegin(ATRACE_TAG, "write"); 2685#endif 2686 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2687#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2688 Tracer::traceEnd(ATRACE_TAG); 2689#endif 2690 if (framesWritten > 0) { 2691 size_t bytesWritten = framesWritten << mBitShift; 2692 mBytesWritten += bytesWritten; 2693 } 2694 2695 mNumWrites++; 2696 mInWrite = false; 2697} 2698 2699void AudioFlinger::MixerThread::threadLoop_standby() 2700{ 2701 // Idle the fast mixer if it's currently running 2702 if (mFastMixer != NULL) { 2703 FastMixerStateQueue *sq = mFastMixer->sq(); 2704 FastMixerState *state = sq->begin(); 2705 if (!(state->mCommand & FastMixerState::IDLE)) { 2706 state->mCommand = FastMixerState::COLD_IDLE; 2707 state->mColdFutexAddr = &mFastMixerFutex; 2708 state->mColdGen++; 2709 mFastMixerFutex = 0; 2710 sq->end(); 2711 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2712 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2713 if (kUseFastMixer == FastMixer_Dynamic) { 2714 mNormalSink = mOutputSink; 2715 } 2716 } else { 2717 sq->end(false /*didModify*/); 2718 } 2719 } 2720 PlaybackThread::threadLoop_standby(); 2721} 2722 2723// shared by MIXER and DIRECT, overridden by DUPLICATING 2724void AudioFlinger::PlaybackThread::threadLoop_standby() 2725{ 2726 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2727 mOutput->stream->common.standby(&mOutput->stream->common); 2728} 2729 2730void AudioFlinger::MixerThread::threadLoop_mix() 2731{ 2732 // obtain the presentation timestamp of the next output buffer 2733 int64_t pts; 2734 status_t status = INVALID_OPERATION; 2735 2736 if (NULL != mOutput->stream->get_next_write_timestamp) { 2737 status = mOutput->stream->get_next_write_timestamp( 2738 mOutput->stream, &pts); 2739 } 2740 2741 if (status != NO_ERROR) { 2742 pts = AudioBufferProvider::kInvalidPTS; 2743 } 2744 2745 // mix buffers... 2746 mAudioMixer->process(pts); 2747 // increase sleep time progressively when application underrun condition clears. 2748 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2749 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2750 // such that we would underrun the audio HAL. 2751 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2752 sleepTimeShift--; 2753 } 2754 sleepTime = 0; 2755 standbyTime = systemTime() + standbyDelay; 2756 //TODO: delay standby when effects have a tail 2757} 2758 2759void AudioFlinger::MixerThread::threadLoop_sleepTime() 2760{ 2761 // If no tracks are ready, sleep once for the duration of an output 2762 // buffer size, then write 0s to the output 2763 if (sleepTime == 0) { 2764 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2765 sleepTime = activeSleepTime >> sleepTimeShift; 2766 if (sleepTime < kMinThreadSleepTimeUs) { 2767 sleepTime = kMinThreadSleepTimeUs; 2768 } 2769 // reduce sleep time in case of consecutive application underruns to avoid 2770 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2771 // duration we would end up writing less data than needed by the audio HAL if 2772 // the condition persists. 2773 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2774 sleepTimeShift++; 2775 } 2776 } else { 2777 sleepTime = idleSleepTime; 2778 } 2779 } else if (mBytesWritten != 0 || 2780 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2781 memset (mMixBuffer, 0, mixBufferSize); 2782 sleepTime = 0; 2783 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2784 } 2785 // TODO add standby time extension fct of effect tail 2786} 2787 2788// prepareTracks_l() must be called with ThreadBase::mLock held 2789AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2790 Vector< sp<Track> > *tracksToRemove) 2791{ 2792 2793 mixer_state mixerStatus = MIXER_IDLE; 2794 // find out which tracks need to be processed 2795 size_t count = mActiveTracks.size(); 2796 size_t mixedTracks = 0; 2797 size_t tracksWithEffect = 0; 2798 // counts only _active_ fast tracks 2799 size_t fastTracks = 0; 2800 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2801 2802 float masterVolume = mMasterVolume; 2803 bool masterMute = mMasterMute; 2804 2805 if (masterMute) { 2806 masterVolume = 0; 2807 } 2808 // Delegate master volume control to effect in output mix effect chain if needed 2809 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2810 if (chain != 0) { 2811 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2812 chain->setVolume_l(&v, &v); 2813 masterVolume = (float)((v + (1 << 23)) >> 24); 2814 chain.clear(); 2815 } 2816 2817 // prepare a new state to push 2818 FastMixerStateQueue *sq = NULL; 2819 FastMixerState *state = NULL; 2820 bool didModify = false; 2821 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2822 if (mFastMixer != NULL) { 2823 sq = mFastMixer->sq(); 2824 state = sq->begin(); 2825 } 2826 2827 for (size_t i=0 ; i<count ; i++) { 2828 sp<Track> t = mActiveTracks[i].promote(); 2829 if (t == 0) continue; 2830 2831 // this const just means the local variable doesn't change 2832 Track* const track = t.get(); 2833 2834 // process fast tracks 2835 if (track->isFastTrack()) { 2836 2837 // It's theoretically possible (though unlikely) for a fast track to be created 2838 // and then removed within the same normal mix cycle. This is not a problem, as 2839 // the track never becomes active so it's fast mixer slot is never touched. 2840 // The converse, of removing an (active) track and then creating a new track 2841 // at the identical fast mixer slot within the same normal mix cycle, 2842 // is impossible because the slot isn't marked available until the end of each cycle. 2843 int j = track->mFastIndex; 2844 FastTrack *fastTrack = &state->mFastTracks[j]; 2845 2846 // Determine whether the track is currently in underrun condition, 2847 // and whether it had a recent underrun. 2848 FastTrackUnderruns underruns = mFastMixerDumpState.mTracks[j].mUnderruns; 2849 uint32_t recentFull = (underruns.mBitFields.mFull - 2850 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2851 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2852 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2853 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2854 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2855 uint32_t recentUnderruns = recentPartial + recentEmpty; 2856 track->mObservedUnderruns = underruns; 2857 // don't count underruns that occur while stopping or pausing 2858 // or stopped which can occur when flush() is called while active 2859 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2860 track->mUnderrunCount += recentUnderruns; 2861 } 2862 2863 // This is similar to the state machine for normal tracks, 2864 // with a few modifications for fast tracks. 2865 bool isActive = true; 2866 switch (track->mState) { 2867 case TrackBase::STOPPING_1: 2868 // track stays active in STOPPING_1 state until first underrun 2869 if (recentUnderruns > 0) { 2870 track->mState = TrackBase::STOPPING_2; 2871 } 2872 break; 2873 case TrackBase::PAUSING: 2874 // ramp down is not yet implemented 2875 track->setPaused(); 2876 break; 2877 case TrackBase::RESUMING: 2878 // ramp up is not yet implemented 2879 track->mState = TrackBase::ACTIVE; 2880 break; 2881 case TrackBase::ACTIVE: 2882 if (recentFull > 0 || recentPartial > 0) { 2883 // track has provided at least some frames recently: reset retry count 2884 track->mRetryCount = kMaxTrackRetries; 2885 } 2886 if (recentUnderruns == 0) { 2887 // no recent underruns: stay active 2888 break; 2889 } 2890 // there has recently been an underrun of some kind 2891 if (track->sharedBuffer() == 0) { 2892 // were any of the recent underruns "empty" (no frames available)? 2893 if (recentEmpty == 0) { 2894 // no, then ignore the partial underruns as they are allowed indefinitely 2895 break; 2896 } 2897 // there has recently been an "empty" underrun: decrement the retry counter 2898 if (--(track->mRetryCount) > 0) { 2899 break; 2900 } 2901 // indicate to client process that the track was disabled because of underrun; 2902 // it will then automatically call start() when data is available 2903 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2904 // remove from active list, but state remains ACTIVE [confusing but true] 2905 isActive = false; 2906 break; 2907 } 2908 // fall through 2909 case TrackBase::STOPPING_2: 2910 case TrackBase::PAUSED: 2911 case TrackBase::TERMINATED: 2912 case TrackBase::STOPPED: 2913 case TrackBase::FLUSHED: // flush() while active 2914 // Check for presentation complete if track is inactive 2915 // We have consumed all the buffers of this track. 2916 // This would be incomplete if we auto-paused on underrun 2917 { 2918 size_t audioHALFrames = 2919 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2920 size_t framesWritten = 2921 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2922 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2923 // track stays in active list until presentation is complete 2924 break; 2925 } 2926 } 2927 if (track->isStopping_2()) { 2928 track->mState = TrackBase::STOPPED; 2929 } 2930 if (track->isStopped()) { 2931 // Can't reset directly, as fast mixer is still polling this track 2932 // track->reset(); 2933 // So instead mark this track as needing to be reset after push with ack 2934 resetMask |= 1 << i; 2935 } 2936 isActive = false; 2937 break; 2938 case TrackBase::IDLE: 2939 default: 2940 LOG_FATAL("unexpected track state %d", track->mState); 2941 } 2942 2943 if (isActive) { 2944 // was it previously inactive? 2945 if (!(state->mTrackMask & (1 << j))) { 2946 ExtendedAudioBufferProvider *eabp = track; 2947 VolumeProvider *vp = track; 2948 fastTrack->mBufferProvider = eabp; 2949 fastTrack->mVolumeProvider = vp; 2950 fastTrack->mSampleRate = track->mSampleRate; 2951 fastTrack->mChannelMask = track->mChannelMask; 2952 fastTrack->mGeneration++; 2953 state->mTrackMask |= 1 << j; 2954 didModify = true; 2955 // no acknowledgement required for newly active tracks 2956 } 2957 // cache the combined master volume and stream type volume for fast mixer; this 2958 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2959 track->mCachedVolume = track->isMuted() ? 2960 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2961 ++fastTracks; 2962 } else { 2963 // was it previously active? 2964 if (state->mTrackMask & (1 << j)) { 2965 fastTrack->mBufferProvider = NULL; 2966 fastTrack->mGeneration++; 2967 state->mTrackMask &= ~(1 << j); 2968 didModify = true; 2969 // If any fast tracks were removed, we must wait for acknowledgement 2970 // because we're about to decrement the last sp<> on those tracks. 2971 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2972 } else { 2973 LOG_FATAL("fast track %d should have been active", j); 2974 } 2975 tracksToRemove->add(track); 2976 // Avoids a misleading display in dumpsys 2977 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2978 } 2979 continue; 2980 } 2981 2982 { // local variable scope to avoid goto warning 2983 2984 audio_track_cblk_t* cblk = track->cblk(); 2985 2986 // The first time a track is added we wait 2987 // for all its buffers to be filled before processing it 2988 int name = track->name(); 2989 // make sure that we have enough frames to mix one full buffer. 2990 // enforce this condition only once to enable draining the buffer in case the client 2991 // app does not call stop() and relies on underrun to stop: 2992 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2993 // during last round 2994 uint32_t minFrames = 1; 2995 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2996 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2997 if (t->sampleRate() == (int)mSampleRate) { 2998 minFrames = mNormalFrameCount; 2999 } else { 3000 // +1 for rounding and +1 for additional sample needed for interpolation 3001 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3002 // add frames already consumed but not yet released by the resampler 3003 // because cblk->framesReady() will include these frames 3004 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3005 // the minimum track buffer size is normally twice the number of frames necessary 3006 // to fill one buffer and the resampler should not leave more than one buffer worth 3007 // of unreleased frames after each pass, but just in case... 3008 ALOG_ASSERT(minFrames <= cblk->frameCount); 3009 } 3010 } 3011 if ((track->framesReady() >= minFrames) && track->isReady() && 3012 !track->isPaused() && !track->isTerminated()) 3013 { 3014 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3015 3016 mixedTracks++; 3017 3018 // track->mainBuffer() != mMixBuffer means there is an effect chain 3019 // connected to the track 3020 chain.clear(); 3021 if (track->mainBuffer() != mMixBuffer) { 3022 chain = getEffectChain_l(track->sessionId()); 3023 // Delegate volume control to effect in track effect chain if needed 3024 if (chain != 0) { 3025 tracksWithEffect++; 3026 } else { 3027 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3028 name, track->sessionId()); 3029 } 3030 } 3031 3032 3033 int param = AudioMixer::VOLUME; 3034 if (track->mFillingUpStatus == Track::FS_FILLED) { 3035 // no ramp for the first volume setting 3036 track->mFillingUpStatus = Track::FS_ACTIVE; 3037 if (track->mState == TrackBase::RESUMING) { 3038 track->mState = TrackBase::ACTIVE; 3039 param = AudioMixer::RAMP_VOLUME; 3040 } 3041 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3042 } else if (cblk->server != 0) { 3043 // If the track is stopped before the first frame was mixed, 3044 // do not apply ramp 3045 param = AudioMixer::RAMP_VOLUME; 3046 } 3047 3048 // compute volume for this track 3049 uint32_t vl, vr, va; 3050 if (track->isMuted() || track->isPausing() || 3051 mStreamTypes[track->streamType()].mute) { 3052 vl = vr = va = 0; 3053 if (track->isPausing()) { 3054 track->setPaused(); 3055 } 3056 } else { 3057 3058 // read original volumes with volume control 3059 float typeVolume = mStreamTypes[track->streamType()].volume; 3060 float v = masterVolume * typeVolume; 3061 uint32_t vlr = cblk->getVolumeLR(); 3062 vl = vlr & 0xFFFF; 3063 vr = vlr >> 16; 3064 // track volumes come from shared memory, so can't be trusted and must be clamped 3065 if (vl > MAX_GAIN_INT) { 3066 ALOGV("Track left volume out of range: %04X", vl); 3067 vl = MAX_GAIN_INT; 3068 } 3069 if (vr > MAX_GAIN_INT) { 3070 ALOGV("Track right volume out of range: %04X", vr); 3071 vr = MAX_GAIN_INT; 3072 } 3073 // now apply the master volume and stream type volume 3074 vl = (uint32_t)(v * vl) << 12; 3075 vr = (uint32_t)(v * vr) << 12; 3076 // assuming master volume and stream type volume each go up to 1.0, 3077 // vl and vr are now in 8.24 format 3078 3079 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3080 // send level comes from shared memory and so may be corrupt 3081 if (sendLevel > MAX_GAIN_INT) { 3082 ALOGV("Track send level out of range: %04X", sendLevel); 3083 sendLevel = MAX_GAIN_INT; 3084 } 3085 va = (uint32_t)(v * sendLevel); 3086 } 3087 // Delegate volume control to effect in track effect chain if needed 3088 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3089 // Do not ramp volume if volume is controlled by effect 3090 param = AudioMixer::VOLUME; 3091 track->mHasVolumeController = true; 3092 } else { 3093 // force no volume ramp when volume controller was just disabled or removed 3094 // from effect chain to avoid volume spike 3095 if (track->mHasVolumeController) { 3096 param = AudioMixer::VOLUME; 3097 } 3098 track->mHasVolumeController = false; 3099 } 3100 3101 // Convert volumes from 8.24 to 4.12 format 3102 // This additional clamping is needed in case chain->setVolume_l() overshot 3103 vl = (vl + (1 << 11)) >> 12; 3104 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3105 vr = (vr + (1 << 11)) >> 12; 3106 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3107 3108 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3109 3110 // XXX: these things DON'T need to be done each time 3111 mAudioMixer->setBufferProvider(name, track); 3112 mAudioMixer->enable(name); 3113 3114 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3115 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3116 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3117 mAudioMixer->setParameter( 3118 name, 3119 AudioMixer::TRACK, 3120 AudioMixer::FORMAT, (void *)track->format()); 3121 mAudioMixer->setParameter( 3122 name, 3123 AudioMixer::TRACK, 3124 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3125 mAudioMixer->setParameter( 3126 name, 3127 AudioMixer::RESAMPLE, 3128 AudioMixer::SAMPLE_RATE, 3129 (void *)(cblk->sampleRate)); 3130 mAudioMixer->setParameter( 3131 name, 3132 AudioMixer::TRACK, 3133 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3134 mAudioMixer->setParameter( 3135 name, 3136 AudioMixer::TRACK, 3137 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3138 3139 // reset retry count 3140 track->mRetryCount = kMaxTrackRetries; 3141 3142 // If one track is ready, set the mixer ready if: 3143 // - the mixer was not ready during previous round OR 3144 // - no other track is not ready 3145 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3146 mixerStatus != MIXER_TRACKS_ENABLED) { 3147 mixerStatus = MIXER_TRACKS_READY; 3148 } 3149 } else { 3150 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3151 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3152 track->isStopped() || track->isPaused()) { 3153 // We have consumed all the buffers of this track. 3154 // Remove it from the list of active tracks. 3155 // TODO: use actual buffer filling status instead of latency when available from 3156 // audio HAL 3157 size_t audioHALFrames = 3158 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3159 size_t framesWritten = 3160 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3161 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3162 if (track->isStopped()) { 3163 track->reset(); 3164 } 3165 tracksToRemove->add(track); 3166 } 3167 } else { 3168 // No buffers for this track. Give it a few chances to 3169 // fill a buffer, then remove it from active list. 3170 if (--(track->mRetryCount) <= 0) { 3171 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3172 tracksToRemove->add(track); 3173 // indicate to client process that the track was disabled because of underrun; 3174 // it will then automatically call start() when data is available 3175 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3176 // If one track is not ready, mark the mixer also not ready if: 3177 // - the mixer was ready during previous round OR 3178 // - no other track is ready 3179 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3180 mixerStatus != MIXER_TRACKS_READY) { 3181 mixerStatus = MIXER_TRACKS_ENABLED; 3182 } 3183 } 3184 mAudioMixer->disable(name); 3185 } 3186 3187 } // local variable scope to avoid goto warning 3188track_is_ready: ; 3189 3190 } 3191 3192 // Push the new FastMixer state if necessary 3193 if (didModify) { 3194 state->mFastTracksGen++; 3195 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3196 if (kUseFastMixer == FastMixer_Dynamic && 3197 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3198 state->mCommand = FastMixerState::COLD_IDLE; 3199 state->mColdFutexAddr = &mFastMixerFutex; 3200 state->mColdGen++; 3201 mFastMixerFutex = 0; 3202 if (kUseFastMixer == FastMixer_Dynamic) { 3203 mNormalSink = mOutputSink; 3204 } 3205 // If we go into cold idle, need to wait for acknowledgement 3206 // so that fast mixer stops doing I/O. 3207 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3208 } 3209 sq->end(); 3210 } 3211 if (sq != NULL) { 3212 sq->end(didModify); 3213 sq->push(block); 3214 } 3215 3216 // Now perform the deferred reset on fast tracks that have stopped 3217 while (resetMask != 0) { 3218 size_t i = __builtin_ctz(resetMask); 3219 ALOG_ASSERT(i < count); 3220 resetMask &= ~(1 << i); 3221 sp<Track> t = mActiveTracks[i].promote(); 3222 if (t == 0) continue; 3223 Track* track = t.get(); 3224 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3225 track->reset(); 3226 } 3227 3228 // remove all the tracks that need to be... 3229 count = tracksToRemove->size(); 3230 if (CC_UNLIKELY(count)) { 3231 for (size_t i=0 ; i<count ; i++) { 3232 const sp<Track>& track = tracksToRemove->itemAt(i); 3233 mActiveTracks.remove(track); 3234 if (track->mainBuffer() != mMixBuffer) { 3235 chain = getEffectChain_l(track->sessionId()); 3236 if (chain != 0) { 3237 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3238 chain->decActiveTrackCnt(); 3239 } 3240 } 3241 if (track->isTerminated()) { 3242 removeTrack_l(track); 3243 } 3244 } 3245 } 3246 3247 // mix buffer must be cleared if all tracks are connected to an 3248 // effect chain as in this case the mixer will not write to 3249 // mix buffer and track effects will accumulate into it 3250 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3251 // FIXME as a performance optimization, should remember previous zero status 3252 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3253 } 3254 3255 // if any fast tracks, then status is ready 3256 mMixerStatusIgnoringFastTracks = mixerStatus; 3257 if (fastTracks > 0) { 3258 mixerStatus = MIXER_TRACKS_READY; 3259 } 3260 return mixerStatus; 3261} 3262 3263/* 3264The derived values that are cached: 3265 - mixBufferSize from frame count * frame size 3266 - activeSleepTime from activeSleepTimeUs() 3267 - idleSleepTime from idleSleepTimeUs() 3268 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3269 - maxPeriod from frame count and sample rate (MIXER only) 3270 3271The parameters that affect these derived values are: 3272 - frame count 3273 - frame size 3274 - sample rate 3275 - device type: A2DP or not 3276 - device latency 3277 - format: PCM or not 3278 - active sleep time 3279 - idle sleep time 3280*/ 3281 3282void AudioFlinger::PlaybackThread::cacheParameters_l() 3283{ 3284 mixBufferSize = mNormalFrameCount * mFrameSize; 3285 activeSleepTime = activeSleepTimeUs(); 3286 idleSleepTime = idleSleepTimeUs(); 3287} 3288 3289void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3290{ 3291 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3292 this, streamType, mTracks.size()); 3293 Mutex::Autolock _l(mLock); 3294 3295 size_t size = mTracks.size(); 3296 for (size_t i = 0; i < size; i++) { 3297 sp<Track> t = mTracks[i]; 3298 if (t->streamType() == streamType) { 3299 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3300 t->mCblk->cv.signal(); 3301 } 3302 } 3303} 3304 3305// getTrackName_l() must be called with ThreadBase::mLock held 3306int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3307{ 3308 return mAudioMixer->getTrackName(channelMask); 3309} 3310 3311// deleteTrackName_l() must be called with ThreadBase::mLock held 3312void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3313{ 3314 ALOGV("remove track (%d) and delete from mixer", name); 3315 mAudioMixer->deleteTrackName(name); 3316} 3317 3318// checkForNewParameters_l() must be called with ThreadBase::mLock held 3319bool AudioFlinger::MixerThread::checkForNewParameters_l() 3320{ 3321 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3322 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3323 bool reconfig = false; 3324 3325 while (!mNewParameters.isEmpty()) { 3326 3327 if (mFastMixer != NULL) { 3328 FastMixerStateQueue *sq = mFastMixer->sq(); 3329 FastMixerState *state = sq->begin(); 3330 if (!(state->mCommand & FastMixerState::IDLE)) { 3331 previousCommand = state->mCommand; 3332 state->mCommand = FastMixerState::HOT_IDLE; 3333 sq->end(); 3334 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3335 } else { 3336 sq->end(false /*didModify*/); 3337 } 3338 } 3339 3340 status_t status = NO_ERROR; 3341 String8 keyValuePair = mNewParameters[0]; 3342 AudioParameter param = AudioParameter(keyValuePair); 3343 int value; 3344 3345 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3346 reconfig = true; 3347 } 3348 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3349 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3350 status = BAD_VALUE; 3351 } else { 3352 reconfig = true; 3353 } 3354 } 3355 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3356 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3357 status = BAD_VALUE; 3358 } else { 3359 reconfig = true; 3360 } 3361 } 3362 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3363 // do not accept frame count changes if tracks are open as the track buffer 3364 // size depends on frame count and correct behavior would not be guaranteed 3365 // if frame count is changed after track creation 3366 if (!mTracks.isEmpty()) { 3367 status = INVALID_OPERATION; 3368 } else { 3369 reconfig = true; 3370 } 3371 } 3372 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3373#ifdef ADD_BATTERY_DATA 3374 // when changing the audio output device, call addBatteryData to notify 3375 // the change 3376 if ((int)mDevice != value) { 3377 uint32_t params = 0; 3378 // check whether speaker is on 3379 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3380 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3381 } 3382 3383 int deviceWithoutSpeaker 3384 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3385 // check if any other device (except speaker) is on 3386 if (value & deviceWithoutSpeaker ) { 3387 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3388 } 3389 3390 if (params != 0) { 3391 addBatteryData(params); 3392 } 3393 } 3394#endif 3395 3396 // forward device change to effects that have requested to be 3397 // aware of attached audio device. 3398 mDevice = (uint32_t)value; 3399 for (size_t i = 0; i < mEffectChains.size(); i++) { 3400 mEffectChains[i]->setDevice_l(mDevice); 3401 } 3402 } 3403 3404 if (status == NO_ERROR) { 3405 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3406 keyValuePair.string()); 3407 if (!mStandby && status == INVALID_OPERATION) { 3408 mOutput->stream->common.standby(&mOutput->stream->common); 3409 mStandby = true; 3410 mBytesWritten = 0; 3411 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3412 keyValuePair.string()); 3413 } 3414 if (status == NO_ERROR && reconfig) { 3415 delete mAudioMixer; 3416 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3417 mAudioMixer = NULL; 3418 readOutputParameters(); 3419 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3420 for (size_t i = 0; i < mTracks.size() ; i++) { 3421 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3422 if (name < 0) break; 3423 mTracks[i]->mName = name; 3424 // limit track sample rate to 2 x new output sample rate 3425 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3426 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3427 } 3428 } 3429 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3430 } 3431 } 3432 3433 mNewParameters.removeAt(0); 3434 3435 mParamStatus = status; 3436 mParamCond.signal(); 3437 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3438 // already timed out waiting for the status and will never signal the condition. 3439 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3440 } 3441 3442 if (!(previousCommand & FastMixerState::IDLE)) { 3443 ALOG_ASSERT(mFastMixer != NULL); 3444 FastMixerStateQueue *sq = mFastMixer->sq(); 3445 FastMixerState *state = sq->begin(); 3446 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3447 state->mCommand = previousCommand; 3448 sq->end(); 3449 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3450 } 3451 3452 return reconfig; 3453} 3454 3455status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3456{ 3457 const size_t SIZE = 256; 3458 char buffer[SIZE]; 3459 String8 result; 3460 3461 PlaybackThread::dumpInternals(fd, args); 3462 3463 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3464 result.append(buffer); 3465 write(fd, result.string(), result.size()); 3466 3467 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3468 FastMixerDumpState copy = mFastMixerDumpState; 3469 copy.dump(fd); 3470 3471 // Write the tee output to a .wav file 3472 NBAIO_Source *teeSource = mTeeSource.get(); 3473 if (teeSource != NULL) { 3474 char teePath[64]; 3475 struct timeval tv; 3476 gettimeofday(&tv, NULL); 3477 struct tm tm; 3478 localtime_r(&tv.tv_sec, &tm); 3479 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3480 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3481 if (teeFd >= 0) { 3482 char wavHeader[44]; 3483 memcpy(wavHeader, 3484 "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", 3485 sizeof(wavHeader)); 3486 NBAIO_Format format = teeSource->format(); 3487 unsigned channelCount = Format_channelCount(format); 3488 ALOG_ASSERT(channelCount <= FCC_2); 3489 unsigned sampleRate = Format_sampleRate(format); 3490 wavHeader[22] = channelCount; // number of channels 3491 wavHeader[24] = sampleRate; // sample rate 3492 wavHeader[25] = sampleRate >> 8; 3493 wavHeader[32] = channelCount * 2; // block alignment 3494 write(teeFd, wavHeader, sizeof(wavHeader)); 3495 size_t total = 0; 3496 bool firstRead = true; 3497 for (;;) { 3498#define TEE_SINK_READ 1024 3499 short buffer[TEE_SINK_READ * FCC_2]; 3500 size_t count = TEE_SINK_READ; 3501 ssize_t actual = teeSource->read(buffer, count); 3502 bool wasFirstRead = firstRead; 3503 firstRead = false; 3504 if (actual <= 0) { 3505 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3506 continue; 3507 } 3508 break; 3509 } 3510 ALOG_ASSERT(actual <= count); 3511 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3512 total += actual; 3513 } 3514 lseek(teeFd, (off_t) 4, SEEK_SET); 3515 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3516 write(teeFd, &temp, sizeof(temp)); 3517 lseek(teeFd, (off_t) 40, SEEK_SET); 3518 temp = total * channelCount * sizeof(short); 3519 write(teeFd, &temp, sizeof(temp)); 3520 close(teeFd); 3521 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3522 } else { 3523 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3524 } 3525 } 3526 3527 return NO_ERROR; 3528} 3529 3530uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3531{ 3532 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3533} 3534 3535uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3536{ 3537 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3538} 3539 3540void AudioFlinger::MixerThread::cacheParameters_l() 3541{ 3542 PlaybackThread::cacheParameters_l(); 3543 3544 // FIXME: Relaxed timing because of a certain device that can't meet latency 3545 // Should be reduced to 2x after the vendor fixes the driver issue 3546 // increase threshold again due to low power audio mode. The way this warning 3547 // threshold is calculated and its usefulness should be reconsidered anyway. 3548 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3549} 3550 3551// ---------------------------------------------------------------------------- 3552AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3553 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3554 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3555 // mLeftVolFloat, mRightVolFloat 3556 // mLeftVolShort, mRightVolShort 3557{ 3558} 3559 3560AudioFlinger::DirectOutputThread::~DirectOutputThread() 3561{ 3562} 3563 3564AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3565 Vector< sp<Track> > *tracksToRemove 3566) 3567{ 3568 sp<Track> trackToRemove; 3569 3570 mixer_state mixerStatus = MIXER_IDLE; 3571 3572 // find out which tracks need to be processed 3573 if (mActiveTracks.size() != 0) { 3574 sp<Track> t = mActiveTracks[0].promote(); 3575 // The track died recently 3576 if (t == 0) return MIXER_IDLE; 3577 3578 Track* const track = t.get(); 3579 audio_track_cblk_t* cblk = track->cblk(); 3580 3581 // The first time a track is added we wait 3582 // for all its buffers to be filled before processing it 3583 if (cblk->framesReady() && track->isReady() && 3584 !track->isPaused() && !track->isTerminated()) 3585 { 3586 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3587 3588 if (track->mFillingUpStatus == Track::FS_FILLED) { 3589 track->mFillingUpStatus = Track::FS_ACTIVE; 3590 mLeftVolFloat = mRightVolFloat = 0; 3591 mLeftVolShort = mRightVolShort = 0; 3592 if (track->mState == TrackBase::RESUMING) { 3593 track->mState = TrackBase::ACTIVE; 3594 rampVolume = true; 3595 } 3596 } else if (cblk->server != 0) { 3597 // If the track is stopped before the first frame was mixed, 3598 // do not apply ramp 3599 rampVolume = true; 3600 } 3601 // compute volume for this track 3602 float left, right; 3603 if (track->isMuted() || mMasterMute || track->isPausing() || 3604 mStreamTypes[track->streamType()].mute) { 3605 left = right = 0; 3606 if (track->isPausing()) { 3607 track->setPaused(); 3608 } 3609 } else { 3610 float typeVolume = mStreamTypes[track->streamType()].volume; 3611 float v = mMasterVolume * typeVolume; 3612 uint32_t vlr = cblk->getVolumeLR(); 3613 float v_clamped = v * (vlr & 0xFFFF); 3614 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3615 left = v_clamped/MAX_GAIN; 3616 v_clamped = v * (vlr >> 16); 3617 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3618 right = v_clamped/MAX_GAIN; 3619 } 3620 3621 if (left != mLeftVolFloat || right != mRightVolFloat) { 3622 mLeftVolFloat = left; 3623 mRightVolFloat = right; 3624 3625 // If audio HAL implements volume control, 3626 // force software volume to nominal value 3627 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3628 left = 1.0f; 3629 right = 1.0f; 3630 } 3631 3632 // Convert volumes from float to 8.24 3633 uint32_t vl = (uint32_t)(left * (1 << 24)); 3634 uint32_t vr = (uint32_t)(right * (1 << 24)); 3635 3636 // Delegate volume control to effect in track effect chain if needed 3637 // only one effect chain can be present on DirectOutputThread, so if 3638 // there is one, the track is connected to it 3639 if (!mEffectChains.isEmpty()) { 3640 // Do not ramp volume if volume is controlled by effect 3641 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3642 rampVolume = false; 3643 } 3644 } 3645 3646 // Convert volumes from 8.24 to 4.12 format 3647 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3648 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3649 leftVol = (uint16_t)v_clamped; 3650 v_clamped = (vr + (1 << 11)) >> 12; 3651 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3652 rightVol = (uint16_t)v_clamped; 3653 } else { 3654 leftVol = mLeftVolShort; 3655 rightVol = mRightVolShort; 3656 rampVolume = false; 3657 } 3658 3659 // reset retry count 3660 track->mRetryCount = kMaxTrackRetriesDirect; 3661 mActiveTrack = t; 3662 mixerStatus = MIXER_TRACKS_READY; 3663 } else { 3664 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3665 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3666 // We have consumed all the buffers of this track. 3667 // Remove it from the list of active tracks. 3668 // TODO: implement behavior for compressed audio 3669 size_t audioHALFrames = 3670 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3671 size_t framesWritten = 3672 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3673 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3674 if (track->isStopped()) { 3675 track->reset(); 3676 } 3677 trackToRemove = track; 3678 } 3679 } else { 3680 // No buffers for this track. Give it a few chances to 3681 // fill a buffer, then remove it from active list. 3682 if (--(track->mRetryCount) <= 0) { 3683 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3684 trackToRemove = track; 3685 } else { 3686 mixerStatus = MIXER_TRACKS_ENABLED; 3687 } 3688 } 3689 } 3690 } 3691 3692 // FIXME merge this with similar code for removing multiple tracks 3693 // remove all the tracks that need to be... 3694 if (CC_UNLIKELY(trackToRemove != 0)) { 3695 tracksToRemove->add(trackToRemove); 3696 mActiveTracks.remove(trackToRemove); 3697 if (!mEffectChains.isEmpty()) { 3698 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3699 trackToRemove->sessionId()); 3700 mEffectChains[0]->decActiveTrackCnt(); 3701 } 3702 if (trackToRemove->isTerminated()) { 3703 removeTrack_l(trackToRemove); 3704 } 3705 } 3706 3707 return mixerStatus; 3708} 3709 3710void AudioFlinger::DirectOutputThread::threadLoop_mix() 3711{ 3712 AudioBufferProvider::Buffer buffer; 3713 size_t frameCount = mFrameCount; 3714 int8_t *curBuf = (int8_t *)mMixBuffer; 3715 // output audio to hardware 3716 while (frameCount) { 3717 buffer.frameCount = frameCount; 3718 mActiveTrack->getNextBuffer(&buffer); 3719 if (CC_UNLIKELY(buffer.raw == NULL)) { 3720 memset(curBuf, 0, frameCount * mFrameSize); 3721 break; 3722 } 3723 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3724 frameCount -= buffer.frameCount; 3725 curBuf += buffer.frameCount * mFrameSize; 3726 mActiveTrack->releaseBuffer(&buffer); 3727 } 3728 sleepTime = 0; 3729 standbyTime = systemTime() + standbyDelay; 3730 mActiveTrack.clear(); 3731 3732 // apply volume 3733 3734 // Do not apply volume on compressed audio 3735 if (!audio_is_linear_pcm(mFormat)) { 3736 return; 3737 } 3738 3739 // convert to signed 16 bit before volume calculation 3740 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3741 size_t count = mFrameCount * mChannelCount; 3742 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3743 int16_t *dst = mMixBuffer + count-1; 3744 while (count--) { 3745 *dst-- = (int16_t)(*src--^0x80) << 8; 3746 } 3747 } 3748 3749 frameCount = mFrameCount; 3750 int16_t *out = mMixBuffer; 3751 if (rampVolume) { 3752 if (mChannelCount == 1) { 3753 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3754 int32_t vlInc = d / (int32_t)frameCount; 3755 int32_t vl = ((int32_t)mLeftVolShort << 16); 3756 do { 3757 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3758 out++; 3759 vl += vlInc; 3760 } while (--frameCount); 3761 3762 } else { 3763 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3764 int32_t vlInc = d / (int32_t)frameCount; 3765 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3766 int32_t vrInc = d / (int32_t)frameCount; 3767 int32_t vl = ((int32_t)mLeftVolShort << 16); 3768 int32_t vr = ((int32_t)mRightVolShort << 16); 3769 do { 3770 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3771 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3772 out += 2; 3773 vl += vlInc; 3774 vr += vrInc; 3775 } while (--frameCount); 3776 } 3777 } else { 3778 if (mChannelCount == 1) { 3779 do { 3780 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3781 out++; 3782 } while (--frameCount); 3783 } else { 3784 do { 3785 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3786 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3787 out += 2; 3788 } while (--frameCount); 3789 } 3790 } 3791 3792 // convert back to unsigned 8 bit after volume calculation 3793 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3794 size_t count = mFrameCount * mChannelCount; 3795 int16_t *src = mMixBuffer; 3796 uint8_t *dst = (uint8_t *)mMixBuffer; 3797 while (count--) { 3798 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3799 } 3800 } 3801 3802 mLeftVolShort = leftVol; 3803 mRightVolShort = rightVol; 3804} 3805 3806void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3807{ 3808 if (sleepTime == 0) { 3809 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3810 sleepTime = activeSleepTime; 3811 } else { 3812 sleepTime = idleSleepTime; 3813 } 3814 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3815 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3816 sleepTime = 0; 3817 } 3818} 3819 3820// getTrackName_l() must be called with ThreadBase::mLock held 3821int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3822{ 3823 return 0; 3824} 3825 3826// deleteTrackName_l() must be called with ThreadBase::mLock held 3827void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3828{ 3829} 3830 3831// checkForNewParameters_l() must be called with ThreadBase::mLock held 3832bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3833{ 3834 bool reconfig = false; 3835 3836 while (!mNewParameters.isEmpty()) { 3837 status_t status = NO_ERROR; 3838 String8 keyValuePair = mNewParameters[0]; 3839 AudioParameter param = AudioParameter(keyValuePair); 3840 int value; 3841 3842 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3843 // do not accept frame count changes if tracks are open as the track buffer 3844 // size depends on frame count and correct behavior would not be garantied 3845 // if frame count is changed after track creation 3846 if (!mTracks.isEmpty()) { 3847 status = INVALID_OPERATION; 3848 } else { 3849 reconfig = true; 3850 } 3851 } 3852 if (status == NO_ERROR) { 3853 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3854 keyValuePair.string()); 3855 if (!mStandby && status == INVALID_OPERATION) { 3856 mOutput->stream->common.standby(&mOutput->stream->common); 3857 mStandby = true; 3858 mBytesWritten = 0; 3859 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3860 keyValuePair.string()); 3861 } 3862 if (status == NO_ERROR && reconfig) { 3863 readOutputParameters(); 3864 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3865 } 3866 } 3867 3868 mNewParameters.removeAt(0); 3869 3870 mParamStatus = status; 3871 mParamCond.signal(); 3872 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3873 // already timed out waiting for the status and will never signal the condition. 3874 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3875 } 3876 return reconfig; 3877} 3878 3879uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3880{ 3881 uint32_t time; 3882 if (audio_is_linear_pcm(mFormat)) { 3883 time = PlaybackThread::activeSleepTimeUs(); 3884 } else { 3885 time = 10000; 3886 } 3887 return time; 3888} 3889 3890uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3891{ 3892 uint32_t time; 3893 if (audio_is_linear_pcm(mFormat)) { 3894 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3895 } else { 3896 time = 10000; 3897 } 3898 return time; 3899} 3900 3901uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3902{ 3903 uint32_t time; 3904 if (audio_is_linear_pcm(mFormat)) { 3905 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3906 } else { 3907 time = 10000; 3908 } 3909 return time; 3910} 3911 3912void AudioFlinger::DirectOutputThread::cacheParameters_l() 3913{ 3914 PlaybackThread::cacheParameters_l(); 3915 3916 // use shorter standby delay as on normal output to release 3917 // hardware resources as soon as possible 3918 standbyDelay = microseconds(activeSleepTime*2); 3919} 3920 3921// ---------------------------------------------------------------------------- 3922 3923AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3924 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3925 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3926 mWaitTimeMs(UINT_MAX) 3927{ 3928 addOutputTrack(mainThread); 3929} 3930 3931AudioFlinger::DuplicatingThread::~DuplicatingThread() 3932{ 3933 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3934 mOutputTracks[i]->destroy(); 3935 } 3936} 3937 3938void AudioFlinger::DuplicatingThread::threadLoop_mix() 3939{ 3940 // mix buffers... 3941 if (outputsReady(outputTracks)) { 3942 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3943 } else { 3944 memset(mMixBuffer, 0, mixBufferSize); 3945 } 3946 sleepTime = 0; 3947 writeFrames = mNormalFrameCount; 3948} 3949 3950void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3951{ 3952 if (sleepTime == 0) { 3953 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3954 sleepTime = activeSleepTime; 3955 } else { 3956 sleepTime = idleSleepTime; 3957 } 3958 } else if (mBytesWritten != 0) { 3959 // flush remaining overflow buffers in output tracks 3960 for (size_t i = 0; i < outputTracks.size(); i++) { 3961 if (outputTracks[i]->isActive()) { 3962 sleepTime = 0; 3963 writeFrames = 0; 3964 memset(mMixBuffer, 0, mixBufferSize); 3965 break; 3966 } 3967 } 3968 } 3969} 3970 3971void AudioFlinger::DuplicatingThread::threadLoop_write() 3972{ 3973 standbyTime = systemTime() + standbyDelay; 3974 for (size_t i = 0; i < outputTracks.size(); i++) { 3975 outputTracks[i]->write(mMixBuffer, writeFrames); 3976 } 3977 mBytesWritten += mixBufferSize; 3978} 3979 3980void AudioFlinger::DuplicatingThread::threadLoop_standby() 3981{ 3982 // DuplicatingThread implements standby by stopping all tracks 3983 for (size_t i = 0; i < outputTracks.size(); i++) { 3984 outputTracks[i]->stop(); 3985 } 3986} 3987 3988void AudioFlinger::DuplicatingThread::saveOutputTracks() 3989{ 3990 outputTracks = mOutputTracks; 3991} 3992 3993void AudioFlinger::DuplicatingThread::clearOutputTracks() 3994{ 3995 outputTracks.clear(); 3996} 3997 3998void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3999{ 4000 Mutex::Autolock _l(mLock); 4001 // FIXME explain this formula 4002 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4003 OutputTrack *outputTrack = new OutputTrack(thread, 4004 this, 4005 mSampleRate, 4006 mFormat, 4007 mChannelMask, 4008 frameCount); 4009 if (outputTrack->cblk() != NULL) { 4010 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4011 mOutputTracks.add(outputTrack); 4012 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4013 updateWaitTime_l(); 4014 } 4015} 4016 4017void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4018{ 4019 Mutex::Autolock _l(mLock); 4020 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4021 if (mOutputTracks[i]->thread() == thread) { 4022 mOutputTracks[i]->destroy(); 4023 mOutputTracks.removeAt(i); 4024 updateWaitTime_l(); 4025 return; 4026 } 4027 } 4028 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4029} 4030 4031// caller must hold mLock 4032void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4033{ 4034 mWaitTimeMs = UINT_MAX; 4035 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4036 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4037 if (strong != 0) { 4038 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4039 if (waitTimeMs < mWaitTimeMs) { 4040 mWaitTimeMs = waitTimeMs; 4041 } 4042 } 4043 } 4044} 4045 4046 4047bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4048{ 4049 for (size_t i = 0; i < outputTracks.size(); i++) { 4050 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4051 if (thread == 0) { 4052 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4053 return false; 4054 } 4055 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4056 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4057 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4058 return false; 4059 } 4060 } 4061 return true; 4062} 4063 4064uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4065{ 4066 return (mWaitTimeMs * 1000) / 2; 4067} 4068 4069void AudioFlinger::DuplicatingThread::cacheParameters_l() 4070{ 4071 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4072 updateWaitTime_l(); 4073 4074 MixerThread::cacheParameters_l(); 4075} 4076 4077// ---------------------------------------------------------------------------- 4078 4079// TrackBase constructor must be called with AudioFlinger::mLock held 4080AudioFlinger::ThreadBase::TrackBase::TrackBase( 4081 ThreadBase *thread, 4082 const sp<Client>& client, 4083 uint32_t sampleRate, 4084 audio_format_t format, 4085 uint32_t channelMask, 4086 int frameCount, 4087 const sp<IMemory>& sharedBuffer, 4088 int sessionId) 4089 : RefBase(), 4090 mThread(thread), 4091 mClient(client), 4092 mCblk(NULL), 4093 // mBuffer 4094 // mBufferEnd 4095 mFrameCount(0), 4096 mState(IDLE), 4097 mSampleRate(sampleRate), 4098 mFormat(format), 4099 mStepServerFailed(false), 4100 mSessionId(sessionId) 4101 // mChannelCount 4102 // mChannelMask 4103{ 4104 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4105 4106 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4107 size_t size = sizeof(audio_track_cblk_t); 4108 uint8_t channelCount = popcount(channelMask); 4109 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4110 if (sharedBuffer == 0) { 4111 size += bufferSize; 4112 } 4113 4114 if (client != NULL) { 4115 mCblkMemory = client->heap()->allocate(size); 4116 if (mCblkMemory != 0) { 4117 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4118 if (mCblk != NULL) { // construct the shared structure in-place. 4119 new(mCblk) audio_track_cblk_t(); 4120 // clear all buffers 4121 mCblk->frameCount = frameCount; 4122 mCblk->sampleRate = sampleRate; 4123// uncomment the following lines to quickly test 32-bit wraparound 4124// mCblk->user = 0xffff0000; 4125// mCblk->server = 0xffff0000; 4126// mCblk->userBase = 0xffff0000; 4127// mCblk->serverBase = 0xffff0000; 4128 mChannelCount = channelCount; 4129 mChannelMask = channelMask; 4130 if (sharedBuffer == 0) { 4131 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4132 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4133 // Force underrun condition to avoid false underrun callback until first data is 4134 // written to buffer (other flags are cleared) 4135 mCblk->flags = CBLK_UNDERRUN_ON; 4136 } else { 4137 mBuffer = sharedBuffer->pointer(); 4138 } 4139 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4140 } 4141 } else { 4142 ALOGE("not enough memory for AudioTrack size=%u", size); 4143 client->heap()->dump("AudioTrack"); 4144 return; 4145 } 4146 } else { 4147 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4148 // construct the shared structure in-place. 4149 new(mCblk) audio_track_cblk_t(); 4150 // clear all buffers 4151 mCblk->frameCount = frameCount; 4152 mCblk->sampleRate = sampleRate; 4153// uncomment the following lines to quickly test 32-bit wraparound 4154// mCblk->user = 0xffff0000; 4155// mCblk->server = 0xffff0000; 4156// mCblk->userBase = 0xffff0000; 4157// mCblk->serverBase = 0xffff0000; 4158 mChannelCount = channelCount; 4159 mChannelMask = channelMask; 4160 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4161 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4162 // Force underrun condition to avoid false underrun callback until first data is 4163 // written to buffer (other flags are cleared) 4164 mCblk->flags = CBLK_UNDERRUN_ON; 4165 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4166 } 4167} 4168 4169AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4170{ 4171 if (mCblk != NULL) { 4172 if (mClient == 0) { 4173 delete mCblk; 4174 } else { 4175 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4176 } 4177 } 4178 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4179 if (mClient != 0) { 4180 // Client destructor must run with AudioFlinger mutex locked 4181 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4182 // If the client's reference count drops to zero, the associated destructor 4183 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4184 // relying on the automatic clear() at end of scope. 4185 mClient.clear(); 4186 } 4187} 4188 4189// AudioBufferProvider interface 4190// getNextBuffer() = 0; 4191// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4192void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4193{ 4194 buffer->raw = NULL; 4195 mFrameCount = buffer->frameCount; 4196 // FIXME See note at getNextBuffer() 4197 (void) step(); // ignore return value of step() 4198 buffer->frameCount = 0; 4199} 4200 4201bool AudioFlinger::ThreadBase::TrackBase::step() { 4202 bool result; 4203 audio_track_cblk_t* cblk = this->cblk(); 4204 4205 result = cblk->stepServer(mFrameCount); 4206 if (!result) { 4207 ALOGV("stepServer failed acquiring cblk mutex"); 4208 mStepServerFailed = true; 4209 } 4210 return result; 4211} 4212 4213void AudioFlinger::ThreadBase::TrackBase::reset() { 4214 audio_track_cblk_t* cblk = this->cblk(); 4215 4216 cblk->user = 0; 4217 cblk->server = 0; 4218 cblk->userBase = 0; 4219 cblk->serverBase = 0; 4220 mStepServerFailed = false; 4221 ALOGV("TrackBase::reset"); 4222} 4223 4224int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4225 return (int)mCblk->sampleRate; 4226} 4227 4228void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4229 audio_track_cblk_t* cblk = this->cblk(); 4230 size_t frameSize = cblk->frameSize; 4231 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4232 int8_t *bufferEnd = bufferStart + frames * frameSize; 4233 4234 // Check validity of returned pointer in case the track control block would have been corrupted. 4235 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4236 "TrackBase::getBuffer buffer out of range:\n" 4237 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4238 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4239 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4240 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4241 4242 return bufferStart; 4243} 4244 4245status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4246{ 4247 mSyncEvents.add(event); 4248 return NO_ERROR; 4249} 4250 4251// ---------------------------------------------------------------------------- 4252 4253// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4254AudioFlinger::PlaybackThread::Track::Track( 4255 PlaybackThread *thread, 4256 const sp<Client>& client, 4257 audio_stream_type_t streamType, 4258 uint32_t sampleRate, 4259 audio_format_t format, 4260 uint32_t channelMask, 4261 int frameCount, 4262 const sp<IMemory>& sharedBuffer, 4263 int sessionId, 4264 IAudioFlinger::track_flags_t flags) 4265 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4266 mMute(false), 4267 mFillingUpStatus(FS_INVALID), 4268 // mRetryCount initialized later when needed 4269 mSharedBuffer(sharedBuffer), 4270 mStreamType(streamType), 4271 mName(-1), // see note below 4272 mMainBuffer(thread->mixBuffer()), 4273 mAuxBuffer(NULL), 4274 mAuxEffectId(0), mHasVolumeController(false), 4275 mPresentationCompleteFrames(0), 4276 mFlags(flags), 4277 mFastIndex(-1), 4278 mUnderrunCount(0), 4279 mCachedVolume(1.0) 4280{ 4281 if (mCblk != NULL) { 4282 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4283 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4284 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4285 if (flags & IAudioFlinger::TRACK_FAST) { 4286 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4287 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4288 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4289 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4290 // FIXME This is too eager. We allocate a fast track index before the 4291 // fast track becomes active. Since fast tracks are a scarce resource, 4292 // this means we are potentially denying other more important fast tracks from 4293 // being created. It would be better to allocate the index dynamically. 4294 mFastIndex = i; 4295 // Read the initial underruns because this field is never cleared by the fast mixer 4296 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4297 thread->mFastTrackAvailMask &= ~(1 << i); 4298 } 4299 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4300 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4301 if (mName < 0) { 4302 ALOGE("no more track names available"); 4303 // FIXME bug - if sufficient fast track indices, but insufficient normal mixer names, 4304 // then we leak a fast track index. Should swap these two sections, or better yet 4305 // only allocate a normal mixer name for normal tracks. 4306 } 4307 } 4308 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4309} 4310 4311AudioFlinger::PlaybackThread::Track::~Track() 4312{ 4313 ALOGV("PlaybackThread::Track destructor"); 4314 sp<ThreadBase> thread = mThread.promote(); 4315 if (thread != 0) { 4316 Mutex::Autolock _l(thread->mLock); 4317 mState = TERMINATED; 4318 } 4319} 4320 4321void AudioFlinger::PlaybackThread::Track::destroy() 4322{ 4323 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4324 // by removing it from mTracks vector, so there is a risk that this Tracks's 4325 // destructor is called. As the destructor needs to lock mLock, 4326 // we must acquire a strong reference on this Track before locking mLock 4327 // here so that the destructor is called only when exiting this function. 4328 // On the other hand, as long as Track::destroy() is only called by 4329 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4330 // this Track with its member mTrack. 4331 sp<Track> keep(this); 4332 { // scope for mLock 4333 sp<ThreadBase> thread = mThread.promote(); 4334 if (thread != 0) { 4335 if (!isOutputTrack()) { 4336 if (mState == ACTIVE || mState == RESUMING) { 4337 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4338 4339#ifdef ADD_BATTERY_DATA 4340 // to track the speaker usage 4341 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4342#endif 4343 } 4344 AudioSystem::releaseOutput(thread->id()); 4345 } 4346 Mutex::Autolock _l(thread->mLock); 4347 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4348 playbackThread->destroyTrack_l(this); 4349 } 4350 } 4351} 4352 4353/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4354{ 4355 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4356 " Server User Main buf Aux Buf Flags FastUnder\n"); 4357} 4358 4359void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4360{ 4361 uint32_t vlr = mCblk->getVolumeLR(); 4362 if (isFastTrack()) { 4363 sprintf(buffer, " F %2d", mFastIndex); 4364 } else { 4365 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4366 } 4367 track_state state = mState; 4368 char stateChar; 4369 switch (state) { 4370 case IDLE: 4371 stateChar = 'I'; 4372 break; 4373 case TERMINATED: 4374 stateChar = 'T'; 4375 break; 4376 case STOPPING_1: 4377 stateChar = 's'; 4378 break; 4379 case STOPPING_2: 4380 stateChar = '5'; 4381 break; 4382 case STOPPED: 4383 stateChar = 'S'; 4384 break; 4385 case RESUMING: 4386 stateChar = 'R'; 4387 break; 4388 case ACTIVE: 4389 stateChar = 'A'; 4390 break; 4391 case PAUSING: 4392 stateChar = 'p'; 4393 break; 4394 case PAUSED: 4395 stateChar = 'P'; 4396 break; 4397 case FLUSHED: 4398 stateChar = 'F'; 4399 break; 4400 default: 4401 stateChar = '?'; 4402 break; 4403 } 4404 char nowInUnderrun; 4405 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4406 case UNDERRUN_FULL: 4407 nowInUnderrun = ' '; 4408 break; 4409 case UNDERRUN_PARTIAL: 4410 nowInUnderrun = '<'; 4411 break; 4412 case UNDERRUN_EMPTY: 4413 nowInUnderrun = '*'; 4414 break; 4415 default: 4416 nowInUnderrun = '?'; 4417 break; 4418 } 4419 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4420 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4421 (mClient == 0) ? getpid_cached : mClient->pid(), 4422 mStreamType, 4423 mFormat, 4424 mChannelMask, 4425 mSessionId, 4426 mFrameCount, 4427 mCblk->frameCount, 4428 stateChar, 4429 mMute, 4430 mFillingUpStatus, 4431 mCblk->sampleRate, 4432 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4433 20.0 * log10((vlr >> 16) / 4096.0), 4434 mCblk->server, 4435 mCblk->user, 4436 (int)mMainBuffer, 4437 (int)mAuxBuffer, 4438 mCblk->flags, 4439 mUnderrunCount, 4440 nowInUnderrun); 4441} 4442 4443// AudioBufferProvider interface 4444status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4445 AudioBufferProvider::Buffer* buffer, int64_t pts) 4446{ 4447 audio_track_cblk_t* cblk = this->cblk(); 4448 uint32_t framesReady; 4449 uint32_t framesReq = buffer->frameCount; 4450 4451 // Check if last stepServer failed, try to step now 4452 if (mStepServerFailed) { 4453 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4454 // Since the fast mixer is higher priority than client callback thread, 4455 // it does not result in priority inversion for client. 4456 // But a non-blocking solution would be preferable to avoid 4457 // fast mixer being unable to tryLock(), and 4458 // to avoid the extra context switches if the client wakes up, 4459 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4460 if (!step()) goto getNextBuffer_exit; 4461 ALOGV("stepServer recovered"); 4462 mStepServerFailed = false; 4463 } 4464 4465 // FIXME Same as above 4466 framesReady = cblk->framesReady(); 4467 4468 if (CC_LIKELY(framesReady)) { 4469 uint32_t s = cblk->server; 4470 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4471 4472 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4473 if (framesReq > framesReady) { 4474 framesReq = framesReady; 4475 } 4476 if (framesReq > bufferEnd - s) { 4477 framesReq = bufferEnd - s; 4478 } 4479 4480 buffer->raw = getBuffer(s, framesReq); 4481 if (buffer->raw == NULL) goto getNextBuffer_exit; 4482 4483 buffer->frameCount = framesReq; 4484 return NO_ERROR; 4485 } 4486 4487getNextBuffer_exit: 4488 buffer->raw = NULL; 4489 buffer->frameCount = 0; 4490 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4491 return NOT_ENOUGH_DATA; 4492} 4493 4494// Note that framesReady() takes a mutex on the control block using tryLock(). 4495// This could result in priority inversion if framesReady() is called by the normal mixer, 4496// as the normal mixer thread runs at lower 4497// priority than the client's callback thread: there is a short window within framesReady() 4498// during which the normal mixer could be preempted, and the client callback would block. 4499// Another problem can occur if framesReady() is called by the fast mixer: 4500// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4501// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4502size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4503 return mCblk->framesReady(); 4504} 4505 4506// Don't call for fast tracks; the framesReady() could result in priority inversion 4507bool AudioFlinger::PlaybackThread::Track::isReady() const { 4508 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4509 4510 if (framesReady() >= mCblk->frameCount || 4511 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4512 mFillingUpStatus = FS_FILLED; 4513 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4514 return true; 4515 } 4516 return false; 4517} 4518 4519status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4520 int triggerSession) 4521{ 4522 status_t status = NO_ERROR; 4523 ALOGV("start(%d), calling pid %d session %d", 4524 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4525 4526 sp<ThreadBase> thread = mThread.promote(); 4527 if (thread != 0) { 4528 Mutex::Autolock _l(thread->mLock); 4529 track_state state = mState; 4530 // here the track could be either new, or restarted 4531 // in both cases "unstop" the track 4532 if (mState == PAUSED) { 4533 mState = TrackBase::RESUMING; 4534 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4535 } else { 4536 mState = TrackBase::ACTIVE; 4537 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4538 } 4539 4540 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4541 thread->mLock.unlock(); 4542 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4543 thread->mLock.lock(); 4544 4545#ifdef ADD_BATTERY_DATA 4546 // to track the speaker usage 4547 if (status == NO_ERROR) { 4548 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4549 } 4550#endif 4551 } 4552 if (status == NO_ERROR) { 4553 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4554 playbackThread->addTrack_l(this); 4555 } else { 4556 mState = state; 4557 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4558 } 4559 } else { 4560 status = BAD_VALUE; 4561 } 4562 return status; 4563} 4564 4565void AudioFlinger::PlaybackThread::Track::stop() 4566{ 4567 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4568 sp<ThreadBase> thread = mThread.promote(); 4569 if (thread != 0) { 4570 Mutex::Autolock _l(thread->mLock); 4571 track_state state = mState; 4572 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4573 // If the track is not active (PAUSED and buffers full), flush buffers 4574 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4575 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4576 reset(); 4577 mState = STOPPED; 4578 } else if (!isFastTrack()) { 4579 mState = STOPPED; 4580 } else { 4581 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4582 // and then to STOPPED and reset() when presentation is complete 4583 mState = STOPPING_1; 4584 } 4585 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4586 } 4587 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4588 thread->mLock.unlock(); 4589 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4590 thread->mLock.lock(); 4591 4592#ifdef ADD_BATTERY_DATA 4593 // to track the speaker usage 4594 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4595#endif 4596 } 4597 } 4598} 4599 4600void AudioFlinger::PlaybackThread::Track::pause() 4601{ 4602 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4603 sp<ThreadBase> thread = mThread.promote(); 4604 if (thread != 0) { 4605 Mutex::Autolock _l(thread->mLock); 4606 if (mState == ACTIVE || mState == RESUMING) { 4607 mState = PAUSING; 4608 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4609 if (!isOutputTrack()) { 4610 thread->mLock.unlock(); 4611 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4612 thread->mLock.lock(); 4613 4614#ifdef ADD_BATTERY_DATA 4615 // to track the speaker usage 4616 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4617#endif 4618 } 4619 } 4620 } 4621} 4622 4623void AudioFlinger::PlaybackThread::Track::flush() 4624{ 4625 ALOGV("flush(%d)", mName); 4626 sp<ThreadBase> thread = mThread.promote(); 4627 if (thread != 0) { 4628 Mutex::Autolock _l(thread->mLock); 4629 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4630 mState != PAUSING) { 4631 return; 4632 } 4633 // No point remaining in PAUSED state after a flush => go to 4634 // FLUSHED state 4635 mState = FLUSHED; 4636 // do not reset the track if it is still in the process of being stopped or paused. 4637 // this will be done by prepareTracks_l() when the track is stopped. 4638 // prepareTracks_l() will see mState == FLUSHED, then 4639 // remove from active track list, reset(), and trigger presentation complete 4640 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4641 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4642 reset(); 4643 } 4644 } 4645} 4646 4647void AudioFlinger::PlaybackThread::Track::reset() 4648{ 4649 // Do not reset twice to avoid discarding data written just after a flush and before 4650 // the audioflinger thread detects the track is stopped. 4651 if (!mResetDone) { 4652 TrackBase::reset(); 4653 // Force underrun condition to avoid false underrun callback until first data is 4654 // written to buffer 4655 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4656 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4657 mFillingUpStatus = FS_FILLING; 4658 mResetDone = true; 4659 if (mState == FLUSHED) { 4660 mState = IDLE; 4661 } 4662 } 4663} 4664 4665void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4666{ 4667 mMute = muted; 4668} 4669 4670status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4671{ 4672 status_t status = DEAD_OBJECT; 4673 sp<ThreadBase> thread = mThread.promote(); 4674 if (thread != 0) { 4675 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4676 status = playbackThread->attachAuxEffect(this, EffectId); 4677 } 4678 return status; 4679} 4680 4681void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4682{ 4683 mAuxEffectId = EffectId; 4684 mAuxBuffer = buffer; 4685} 4686 4687bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4688 size_t audioHalFrames) 4689{ 4690 // a track is considered presented when the total number of frames written to audio HAL 4691 // corresponds to the number of frames written when presentationComplete() is called for the 4692 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4693 if (mPresentationCompleteFrames == 0) { 4694 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4695 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4696 mPresentationCompleteFrames, audioHalFrames); 4697 } 4698 if (framesWritten >= mPresentationCompleteFrames) { 4699 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4700 mSessionId, framesWritten); 4701 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4702 return true; 4703 } 4704 return false; 4705} 4706 4707void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4708{ 4709 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4710 if (mSyncEvents[i]->type() == type) { 4711 mSyncEvents[i]->trigger(); 4712 mSyncEvents.removeAt(i); 4713 i--; 4714 } 4715 } 4716} 4717 4718// implement VolumeBufferProvider interface 4719 4720uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4721{ 4722 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4723 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4724 uint32_t vlr = mCblk->getVolumeLR(); 4725 uint32_t vl = vlr & 0xFFFF; 4726 uint32_t vr = vlr >> 16; 4727 // track volumes come from shared memory, so can't be trusted and must be clamped 4728 if (vl > MAX_GAIN_INT) { 4729 vl = MAX_GAIN_INT; 4730 } 4731 if (vr > MAX_GAIN_INT) { 4732 vr = MAX_GAIN_INT; 4733 } 4734 // now apply the cached master volume and stream type volume; 4735 // this is trusted but lacks any synchronization or barrier so may be stale 4736 float v = mCachedVolume; 4737 vl *= v; 4738 vr *= v; 4739 // re-combine into U4.16 4740 vlr = (vr << 16) | (vl & 0xFFFF); 4741 // FIXME look at mute, pause, and stop flags 4742 return vlr; 4743} 4744 4745status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4746{ 4747 if (mState == TERMINATED || mState == PAUSED || 4748 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4749 (mState == STOPPED)))) { 4750 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4751 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4752 event->cancel(); 4753 return INVALID_OPERATION; 4754 } 4755 TrackBase::setSyncEvent(event); 4756 return NO_ERROR; 4757} 4758 4759// timed audio tracks 4760 4761sp<AudioFlinger::PlaybackThread::TimedTrack> 4762AudioFlinger::PlaybackThread::TimedTrack::create( 4763 PlaybackThread *thread, 4764 const sp<Client>& client, 4765 audio_stream_type_t streamType, 4766 uint32_t sampleRate, 4767 audio_format_t format, 4768 uint32_t channelMask, 4769 int frameCount, 4770 const sp<IMemory>& sharedBuffer, 4771 int sessionId) { 4772 if (!client->reserveTimedTrack()) 4773 return NULL; 4774 4775 return new TimedTrack( 4776 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4777 sharedBuffer, sessionId); 4778} 4779 4780AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4781 PlaybackThread *thread, 4782 const sp<Client>& client, 4783 audio_stream_type_t streamType, 4784 uint32_t sampleRate, 4785 audio_format_t format, 4786 uint32_t channelMask, 4787 int frameCount, 4788 const sp<IMemory>& sharedBuffer, 4789 int sessionId) 4790 : Track(thread, client, streamType, sampleRate, format, channelMask, 4791 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4792 mQueueHeadInFlight(false), 4793 mTrimQueueHeadOnRelease(false), 4794 mFramesPendingInQueue(0), 4795 mTimedSilenceBuffer(NULL), 4796 mTimedSilenceBufferSize(0), 4797 mTimedAudioOutputOnTime(false), 4798 mMediaTimeTransformValid(false) 4799{ 4800 LocalClock lc; 4801 mLocalTimeFreq = lc.getLocalFreq(); 4802 4803 mLocalTimeToSampleTransform.a_zero = 0; 4804 mLocalTimeToSampleTransform.b_zero = 0; 4805 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4806 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4807 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4808 &mLocalTimeToSampleTransform.a_to_b_denom); 4809 4810 mMediaTimeToSampleTransform.a_zero = 0; 4811 mMediaTimeToSampleTransform.b_zero = 0; 4812 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4813 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4814 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4815 &mMediaTimeToSampleTransform.a_to_b_denom); 4816} 4817 4818AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4819 mClient->releaseTimedTrack(); 4820 delete [] mTimedSilenceBuffer; 4821} 4822 4823status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4824 size_t size, sp<IMemory>* buffer) { 4825 4826 Mutex::Autolock _l(mTimedBufferQueueLock); 4827 4828 trimTimedBufferQueue_l(); 4829 4830 // lazily initialize the shared memory heap for timed buffers 4831 if (mTimedMemoryDealer == NULL) { 4832 const int kTimedBufferHeapSize = 512 << 10; 4833 4834 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4835 "AudioFlingerTimed"); 4836 if (mTimedMemoryDealer == NULL) 4837 return NO_MEMORY; 4838 } 4839 4840 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4841 if (newBuffer == NULL) { 4842 newBuffer = mTimedMemoryDealer->allocate(size); 4843 if (newBuffer == NULL) 4844 return NO_MEMORY; 4845 } 4846 4847 *buffer = newBuffer; 4848 return NO_ERROR; 4849} 4850 4851// caller must hold mTimedBufferQueueLock 4852void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4853 int64_t mediaTimeNow; 4854 { 4855 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4856 if (!mMediaTimeTransformValid) 4857 return; 4858 4859 int64_t targetTimeNow; 4860 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4861 ? mCCHelper.getCommonTime(&targetTimeNow) 4862 : mCCHelper.getLocalTime(&targetTimeNow); 4863 4864 if (OK != res) 4865 return; 4866 4867 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4868 &mediaTimeNow)) { 4869 return; 4870 } 4871 } 4872 4873 size_t trimEnd; 4874 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4875 int64_t bufEnd; 4876 4877 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4878 // We have a next buffer. Just use its PTS as the PTS of the frame 4879 // following the last frame in this buffer. If the stream is sparse 4880 // (ie, there are deliberate gaps left in the stream which should be 4881 // filled with silence by the TimedAudioTrack), then this can result 4882 // in one extra buffer being left un-trimmed when it could have 4883 // been. In general, this is not typical, and we would rather 4884 // optimized away the TS calculation below for the more common case 4885 // where PTSes are contiguous. 4886 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4887 } else { 4888 // We have no next buffer. Compute the PTS of the frame following 4889 // the last frame in this buffer by computing the duration of of 4890 // this frame in media time units and adding it to the PTS of the 4891 // buffer. 4892 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4893 / mCblk->frameSize; 4894 4895 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4896 &bufEnd)) { 4897 ALOGE("Failed to convert frame count of %lld to media time" 4898 " duration" " (scale factor %d/%u) in %s", 4899 frameCount, 4900 mMediaTimeToSampleTransform.a_to_b_numer, 4901 mMediaTimeToSampleTransform.a_to_b_denom, 4902 __PRETTY_FUNCTION__); 4903 break; 4904 } 4905 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4906 } 4907 4908 if (bufEnd > mediaTimeNow) 4909 break; 4910 4911 // Is the buffer we want to use in the middle of a mix operation right 4912 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4913 // from the mixer which should be coming back shortly. 4914 if (!trimEnd && mQueueHeadInFlight) { 4915 mTrimQueueHeadOnRelease = true; 4916 } 4917 } 4918 4919 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4920 if (trimStart < trimEnd) { 4921 // Update the bookkeeping for framesReady() 4922 for (size_t i = trimStart; i < trimEnd; ++i) { 4923 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4924 } 4925 4926 // Now actually remove the buffers from the queue. 4927 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4928 } 4929} 4930 4931void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4932 const char* logTag) { 4933 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4934 "%s called (reason \"%s\"), but timed buffer queue has no" 4935 " elements to trim.", __FUNCTION__, logTag); 4936 4937 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4938 mTimedBufferQueue.removeAt(0); 4939} 4940 4941void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4942 const TimedBuffer& buf, 4943 const char* logTag) { 4944 uint32_t bufBytes = buf.buffer()->size(); 4945 uint32_t consumedAlready = buf.position(); 4946 4947 ALOG_ASSERT(consumedAlready <= bufBytes, 4948 "Bad bookkeeping while updating frames pending. Timed buffer is" 4949 " only %u bytes long, but claims to have consumed %u" 4950 " bytes. (update reason: \"%s\")", 4951 bufBytes, consumedAlready, logTag); 4952 4953 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4954 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4955 "Bad bookkeeping while updating frames pending. Should have at" 4956 " least %u queued frames, but we think we have only %u. (update" 4957 " reason: \"%s\")", 4958 bufFrames, mFramesPendingInQueue, logTag); 4959 4960 mFramesPendingInQueue -= bufFrames; 4961} 4962 4963status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4964 const sp<IMemory>& buffer, int64_t pts) { 4965 4966 { 4967 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4968 if (!mMediaTimeTransformValid) 4969 return INVALID_OPERATION; 4970 } 4971 4972 Mutex::Autolock _l(mTimedBufferQueueLock); 4973 4974 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4975 mFramesPendingInQueue += bufFrames; 4976 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4977 4978 return NO_ERROR; 4979} 4980 4981status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4982 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4983 4984 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4985 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4986 target); 4987 4988 if (!(target == TimedAudioTrack::LOCAL_TIME || 4989 target == TimedAudioTrack::COMMON_TIME)) { 4990 return BAD_VALUE; 4991 } 4992 4993 Mutex::Autolock lock(mMediaTimeTransformLock); 4994 mMediaTimeTransform = xform; 4995 mMediaTimeTransformTarget = target; 4996 mMediaTimeTransformValid = true; 4997 4998 return NO_ERROR; 4999} 5000 5001#define min(a, b) ((a) < (b) ? (a) : (b)) 5002 5003// implementation of getNextBuffer for tracks whose buffers have timestamps 5004status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5005 AudioBufferProvider::Buffer* buffer, int64_t pts) 5006{ 5007 if (pts == AudioBufferProvider::kInvalidPTS) { 5008 buffer->raw = 0; 5009 buffer->frameCount = 0; 5010 mTimedAudioOutputOnTime = false; 5011 return INVALID_OPERATION; 5012 } 5013 5014 Mutex::Autolock _l(mTimedBufferQueueLock); 5015 5016 ALOG_ASSERT(!mQueueHeadInFlight, 5017 "getNextBuffer called without releaseBuffer!"); 5018 5019 while (true) { 5020 5021 // if we have no timed buffers, then fail 5022 if (mTimedBufferQueue.isEmpty()) { 5023 buffer->raw = 0; 5024 buffer->frameCount = 0; 5025 return NOT_ENOUGH_DATA; 5026 } 5027 5028 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5029 5030 // calculate the PTS of the head of the timed buffer queue expressed in 5031 // local time 5032 int64_t headLocalPTS; 5033 { 5034 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5035 5036 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5037 5038 if (mMediaTimeTransform.a_to_b_denom == 0) { 5039 // the transform represents a pause, so yield silence 5040 timedYieldSilence_l(buffer->frameCount, buffer); 5041 return NO_ERROR; 5042 } 5043 5044 int64_t transformedPTS; 5045 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5046 &transformedPTS)) { 5047 // the transform failed. this shouldn't happen, but if it does 5048 // then just drop this buffer 5049 ALOGW("timedGetNextBuffer transform failed"); 5050 buffer->raw = 0; 5051 buffer->frameCount = 0; 5052 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5053 return NO_ERROR; 5054 } 5055 5056 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5057 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5058 &headLocalPTS)) { 5059 buffer->raw = 0; 5060 buffer->frameCount = 0; 5061 return INVALID_OPERATION; 5062 } 5063 } else { 5064 headLocalPTS = transformedPTS; 5065 } 5066 } 5067 5068 // adjust the head buffer's PTS to reflect the portion of the head buffer 5069 // that has already been consumed 5070 int64_t effectivePTS = headLocalPTS + 5071 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5072 5073 // Calculate the delta in samples between the head of the input buffer 5074 // queue and the start of the next output buffer that will be written. 5075 // If the transformation fails because of over or underflow, it means 5076 // that the sample's position in the output stream is so far out of 5077 // whack that it should just be dropped. 5078 int64_t sampleDelta; 5079 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5080 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5081 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5082 " mix"); 5083 continue; 5084 } 5085 if (!mLocalTimeToSampleTransform.doForwardTransform( 5086 (effectivePTS - pts) << 32, &sampleDelta)) { 5087 ALOGV("*** too late during sample rate transform: dropped buffer"); 5088 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5089 continue; 5090 } 5091 5092 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5093 " sampleDelta=[%d.%08x]", 5094 head.pts(), head.position(), pts, 5095 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5096 + (sampleDelta >> 32)), 5097 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5098 5099 // if the delta between the ideal placement for the next input sample and 5100 // the current output position is within this threshold, then we will 5101 // concatenate the next input samples to the previous output 5102 const int64_t kSampleContinuityThreshold = 5103 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5104 5105 // if this is the first buffer of audio that we're emitting from this track 5106 // then it should be almost exactly on time. 5107 const int64_t kSampleStartupThreshold = 1LL << 32; 5108 5109 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5110 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5111 // the next input is close enough to being on time, so concatenate it 5112 // with the last output 5113 timedYieldSamples_l(buffer); 5114 5115 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5116 head.position(), buffer->frameCount); 5117 return NO_ERROR; 5118 } 5119 5120 // Looks like our output is not on time. Reset our on timed status. 5121 // Next time we mix samples from our input queue, then should be within 5122 // the StartupThreshold. 5123 mTimedAudioOutputOnTime = false; 5124 if (sampleDelta > 0) { 5125 // the gap between the current output position and the proper start of 5126 // the next input sample is too big, so fill it with silence 5127 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5128 5129 timedYieldSilence_l(framesUntilNextInput, buffer); 5130 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5131 return NO_ERROR; 5132 } else { 5133 // the next input sample is late 5134 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5135 size_t onTimeSamplePosition = 5136 head.position() + lateFrames * mCblk->frameSize; 5137 5138 if (onTimeSamplePosition > head.buffer()->size()) { 5139 // all the remaining samples in the head are too late, so 5140 // drop it and move on 5141 ALOGV("*** too late: dropped buffer"); 5142 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5143 continue; 5144 } else { 5145 // skip over the late samples 5146 head.setPosition(onTimeSamplePosition); 5147 5148 // yield the available samples 5149 timedYieldSamples_l(buffer); 5150 5151 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5152 return NO_ERROR; 5153 } 5154 } 5155 } 5156} 5157 5158// Yield samples from the timed buffer queue head up to the given output 5159// buffer's capacity. 5160// 5161// Caller must hold mTimedBufferQueueLock 5162void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5163 AudioBufferProvider::Buffer* buffer) { 5164 5165 const TimedBuffer& head = mTimedBufferQueue[0]; 5166 5167 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5168 head.position()); 5169 5170 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5171 mCblk->frameSize); 5172 size_t framesRequested = buffer->frameCount; 5173 buffer->frameCount = min(framesLeftInHead, framesRequested); 5174 5175 mQueueHeadInFlight = true; 5176 mTimedAudioOutputOnTime = true; 5177} 5178 5179// Yield samples of silence up to the given output buffer's capacity 5180// 5181// Caller must hold mTimedBufferQueueLock 5182void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5183 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5184 5185 // lazily allocate a buffer filled with silence 5186 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5187 delete [] mTimedSilenceBuffer; 5188 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5189 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5190 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5191 } 5192 5193 buffer->raw = mTimedSilenceBuffer; 5194 size_t framesRequested = buffer->frameCount; 5195 buffer->frameCount = min(numFrames, framesRequested); 5196 5197 mTimedAudioOutputOnTime = false; 5198} 5199 5200// AudioBufferProvider interface 5201void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5202 AudioBufferProvider::Buffer* buffer) { 5203 5204 Mutex::Autolock _l(mTimedBufferQueueLock); 5205 5206 // If the buffer which was just released is part of the buffer at the head 5207 // of the queue, be sure to update the amt of the buffer which has been 5208 // consumed. If the buffer being returned is not part of the head of the 5209 // queue, its either because the buffer is part of the silence buffer, or 5210 // because the head of the timed queue was trimmed after the mixer called 5211 // getNextBuffer but before the mixer called releaseBuffer. 5212 if (buffer->raw == mTimedSilenceBuffer) { 5213 ALOG_ASSERT(!mQueueHeadInFlight, 5214 "Queue head in flight during release of silence buffer!"); 5215 goto done; 5216 } 5217 5218 ALOG_ASSERT(mQueueHeadInFlight, 5219 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5220 " head in flight."); 5221 5222 if (mTimedBufferQueue.size()) { 5223 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5224 5225 void* start = head.buffer()->pointer(); 5226 void* end = reinterpret_cast<void*>( 5227 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5228 + head.buffer()->size()); 5229 5230 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5231 "released buffer not within the head of the timed buffer" 5232 " queue; qHead = [%p, %p], released buffer = %p", 5233 start, end, buffer->raw); 5234 5235 head.setPosition(head.position() + 5236 (buffer->frameCount * mCblk->frameSize)); 5237 mQueueHeadInFlight = false; 5238 5239 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5240 "Bad bookkeeping during releaseBuffer! Should have at" 5241 " least %u queued frames, but we think we have only %u", 5242 buffer->frameCount, mFramesPendingInQueue); 5243 5244 mFramesPendingInQueue -= buffer->frameCount; 5245 5246 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5247 || mTrimQueueHeadOnRelease) { 5248 trimTimedBufferQueueHead_l("releaseBuffer"); 5249 mTrimQueueHeadOnRelease = false; 5250 } 5251 } else { 5252 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5253 " buffers in the timed buffer queue"); 5254 } 5255 5256done: 5257 buffer->raw = 0; 5258 buffer->frameCount = 0; 5259} 5260 5261size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5262 Mutex::Autolock _l(mTimedBufferQueueLock); 5263 return mFramesPendingInQueue; 5264} 5265 5266AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5267 : mPTS(0), mPosition(0) {} 5268 5269AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5270 const sp<IMemory>& buffer, int64_t pts) 5271 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5272 5273// ---------------------------------------------------------------------------- 5274 5275// RecordTrack constructor must be called with AudioFlinger::mLock held 5276AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5277 RecordThread *thread, 5278 const sp<Client>& client, 5279 uint32_t sampleRate, 5280 audio_format_t format, 5281 uint32_t channelMask, 5282 int frameCount, 5283 int sessionId) 5284 : TrackBase(thread, client, sampleRate, format, 5285 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5286 mOverflow(false) 5287{ 5288 if (mCblk != NULL) { 5289 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5290 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5291 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5292 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5293 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5294 } else { 5295 mCblk->frameSize = sizeof(int8_t); 5296 } 5297 } 5298} 5299 5300AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5301{ 5302 sp<ThreadBase> thread = mThread.promote(); 5303 if (thread != 0) { 5304 AudioSystem::releaseInput(thread->id()); 5305 } 5306} 5307 5308// AudioBufferProvider interface 5309status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5310{ 5311 audio_track_cblk_t* cblk = this->cblk(); 5312 uint32_t framesAvail; 5313 uint32_t framesReq = buffer->frameCount; 5314 5315 // Check if last stepServer failed, try to step now 5316 if (mStepServerFailed) { 5317 if (!step()) goto getNextBuffer_exit; 5318 ALOGV("stepServer recovered"); 5319 mStepServerFailed = false; 5320 } 5321 5322 framesAvail = cblk->framesAvailable_l(); 5323 5324 if (CC_LIKELY(framesAvail)) { 5325 uint32_t s = cblk->server; 5326 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5327 5328 if (framesReq > framesAvail) { 5329 framesReq = framesAvail; 5330 } 5331 if (framesReq > bufferEnd - s) { 5332 framesReq = bufferEnd - s; 5333 } 5334 5335 buffer->raw = getBuffer(s, framesReq); 5336 if (buffer->raw == NULL) goto getNextBuffer_exit; 5337 5338 buffer->frameCount = framesReq; 5339 return NO_ERROR; 5340 } 5341 5342getNextBuffer_exit: 5343 buffer->raw = NULL; 5344 buffer->frameCount = 0; 5345 return NOT_ENOUGH_DATA; 5346} 5347 5348status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5349 int triggerSession) 5350{ 5351 sp<ThreadBase> thread = mThread.promote(); 5352 if (thread != 0) { 5353 RecordThread *recordThread = (RecordThread *)thread.get(); 5354 return recordThread->start(this, event, triggerSession); 5355 } else { 5356 return BAD_VALUE; 5357 } 5358} 5359 5360void AudioFlinger::RecordThread::RecordTrack::stop() 5361{ 5362 sp<ThreadBase> thread = mThread.promote(); 5363 if (thread != 0) { 5364 RecordThread *recordThread = (RecordThread *)thread.get(); 5365 recordThread->stop(this); 5366 TrackBase::reset(); 5367 // Force overrun condition to avoid false overrun callback until first data is 5368 // read from buffer 5369 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5370 } 5371} 5372 5373void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5374{ 5375 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5376 (mClient == 0) ? getpid_cached : mClient->pid(), 5377 mFormat, 5378 mChannelMask, 5379 mSessionId, 5380 mFrameCount, 5381 mState, 5382 mCblk->sampleRate, 5383 mCblk->server, 5384 mCblk->user); 5385} 5386 5387 5388// ---------------------------------------------------------------------------- 5389 5390AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5391 PlaybackThread *playbackThread, 5392 DuplicatingThread *sourceThread, 5393 uint32_t sampleRate, 5394 audio_format_t format, 5395 uint32_t channelMask, 5396 int frameCount) 5397 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5398 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5399 mActive(false), mSourceThread(sourceThread) 5400{ 5401 5402 if (mCblk != NULL) { 5403 mCblk->flags |= CBLK_DIRECTION_OUT; 5404 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5405 mOutBuffer.frameCount = 0; 5406 playbackThread->mTracks.add(this); 5407 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5408 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5409 mCblk, mBuffer, mCblk->buffers, 5410 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5411 } else { 5412 ALOGW("Error creating output track on thread %p", playbackThread); 5413 } 5414} 5415 5416AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5417{ 5418 clearBufferQueue(); 5419} 5420 5421status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5422 int triggerSession) 5423{ 5424 status_t status = Track::start(event, triggerSession); 5425 if (status != NO_ERROR) { 5426 return status; 5427 } 5428 5429 mActive = true; 5430 mRetryCount = 127; 5431 return status; 5432} 5433 5434void AudioFlinger::PlaybackThread::OutputTrack::stop() 5435{ 5436 Track::stop(); 5437 clearBufferQueue(); 5438 mOutBuffer.frameCount = 0; 5439 mActive = false; 5440} 5441 5442bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5443{ 5444 Buffer *pInBuffer; 5445 Buffer inBuffer; 5446 uint32_t channelCount = mChannelCount; 5447 bool outputBufferFull = false; 5448 inBuffer.frameCount = frames; 5449 inBuffer.i16 = data; 5450 5451 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5452 5453 if (!mActive && frames != 0) { 5454 start(); 5455 sp<ThreadBase> thread = mThread.promote(); 5456 if (thread != 0) { 5457 MixerThread *mixerThread = (MixerThread *)thread.get(); 5458 if (mCblk->frameCount > frames){ 5459 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5460 uint32_t startFrames = (mCblk->frameCount - frames); 5461 pInBuffer = new Buffer; 5462 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5463 pInBuffer->frameCount = startFrames; 5464 pInBuffer->i16 = pInBuffer->mBuffer; 5465 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5466 mBufferQueue.add(pInBuffer); 5467 } else { 5468 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5469 } 5470 } 5471 } 5472 } 5473 5474 while (waitTimeLeftMs) { 5475 // First write pending buffers, then new data 5476 if (mBufferQueue.size()) { 5477 pInBuffer = mBufferQueue.itemAt(0); 5478 } else { 5479 pInBuffer = &inBuffer; 5480 } 5481 5482 if (pInBuffer->frameCount == 0) { 5483 break; 5484 } 5485 5486 if (mOutBuffer.frameCount == 0) { 5487 mOutBuffer.frameCount = pInBuffer->frameCount; 5488 nsecs_t startTime = systemTime(); 5489 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5490 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5491 outputBufferFull = true; 5492 break; 5493 } 5494 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5495 if (waitTimeLeftMs >= waitTimeMs) { 5496 waitTimeLeftMs -= waitTimeMs; 5497 } else { 5498 waitTimeLeftMs = 0; 5499 } 5500 } 5501 5502 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5503 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5504 mCblk->stepUser(outFrames); 5505 pInBuffer->frameCount -= outFrames; 5506 pInBuffer->i16 += outFrames * channelCount; 5507 mOutBuffer.frameCount -= outFrames; 5508 mOutBuffer.i16 += outFrames * channelCount; 5509 5510 if (pInBuffer->frameCount == 0) { 5511 if (mBufferQueue.size()) { 5512 mBufferQueue.removeAt(0); 5513 delete [] pInBuffer->mBuffer; 5514 delete pInBuffer; 5515 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5516 } else { 5517 break; 5518 } 5519 } 5520 } 5521 5522 // If we could not write all frames, allocate a buffer and queue it for next time. 5523 if (inBuffer.frameCount) { 5524 sp<ThreadBase> thread = mThread.promote(); 5525 if (thread != 0 && !thread->standby()) { 5526 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5527 pInBuffer = new Buffer; 5528 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5529 pInBuffer->frameCount = inBuffer.frameCount; 5530 pInBuffer->i16 = pInBuffer->mBuffer; 5531 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5532 mBufferQueue.add(pInBuffer); 5533 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5534 } else { 5535 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5536 } 5537 } 5538 } 5539 5540 // Calling write() with a 0 length buffer, means that no more data will be written: 5541 // If no more buffers are pending, fill output track buffer to make sure it is started 5542 // by output mixer. 5543 if (frames == 0 && mBufferQueue.size() == 0) { 5544 if (mCblk->user < mCblk->frameCount) { 5545 frames = mCblk->frameCount - mCblk->user; 5546 pInBuffer = new Buffer; 5547 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5548 pInBuffer->frameCount = frames; 5549 pInBuffer->i16 = pInBuffer->mBuffer; 5550 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5551 mBufferQueue.add(pInBuffer); 5552 } else if (mActive) { 5553 stop(); 5554 } 5555 } 5556 5557 return outputBufferFull; 5558} 5559 5560status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5561{ 5562 int active; 5563 status_t result; 5564 audio_track_cblk_t* cblk = mCblk; 5565 uint32_t framesReq = buffer->frameCount; 5566 5567// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5568 buffer->frameCount = 0; 5569 5570 uint32_t framesAvail = cblk->framesAvailable(); 5571 5572 5573 if (framesAvail == 0) { 5574 Mutex::Autolock _l(cblk->lock); 5575 goto start_loop_here; 5576 while (framesAvail == 0) { 5577 active = mActive; 5578 if (CC_UNLIKELY(!active)) { 5579 ALOGV("Not active and NO_MORE_BUFFERS"); 5580 return NO_MORE_BUFFERS; 5581 } 5582 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5583 if (result != NO_ERROR) { 5584 return NO_MORE_BUFFERS; 5585 } 5586 // read the server count again 5587 start_loop_here: 5588 framesAvail = cblk->framesAvailable_l(); 5589 } 5590 } 5591 5592// if (framesAvail < framesReq) { 5593// return NO_MORE_BUFFERS; 5594// } 5595 5596 if (framesReq > framesAvail) { 5597 framesReq = framesAvail; 5598 } 5599 5600 uint32_t u = cblk->user; 5601 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5602 5603 if (framesReq > bufferEnd - u) { 5604 framesReq = bufferEnd - u; 5605 } 5606 5607 buffer->frameCount = framesReq; 5608 buffer->raw = (void *)cblk->buffer(u); 5609 return NO_ERROR; 5610} 5611 5612 5613void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5614{ 5615 size_t size = mBufferQueue.size(); 5616 5617 for (size_t i = 0; i < size; i++) { 5618 Buffer *pBuffer = mBufferQueue.itemAt(i); 5619 delete [] pBuffer->mBuffer; 5620 delete pBuffer; 5621 } 5622 mBufferQueue.clear(); 5623} 5624 5625// ---------------------------------------------------------------------------- 5626 5627AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5628 : RefBase(), 5629 mAudioFlinger(audioFlinger), 5630 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5631 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5632 mPid(pid), 5633 mTimedTrackCount(0) 5634{ 5635 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5636} 5637 5638// Client destructor must be called with AudioFlinger::mLock held 5639AudioFlinger::Client::~Client() 5640{ 5641 mAudioFlinger->removeClient_l(mPid); 5642} 5643 5644sp<MemoryDealer> AudioFlinger::Client::heap() const 5645{ 5646 return mMemoryDealer; 5647} 5648 5649// Reserve one of the limited slots for a timed audio track associated 5650// with this client 5651bool AudioFlinger::Client::reserveTimedTrack() 5652{ 5653 const int kMaxTimedTracksPerClient = 4; 5654 5655 Mutex::Autolock _l(mTimedTrackLock); 5656 5657 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5658 ALOGW("can not create timed track - pid %d has exceeded the limit", 5659 mPid); 5660 return false; 5661 } 5662 5663 mTimedTrackCount++; 5664 return true; 5665} 5666 5667// Release a slot for a timed audio track 5668void AudioFlinger::Client::releaseTimedTrack() 5669{ 5670 Mutex::Autolock _l(mTimedTrackLock); 5671 mTimedTrackCount--; 5672} 5673 5674// ---------------------------------------------------------------------------- 5675 5676AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5677 const sp<IAudioFlingerClient>& client, 5678 pid_t pid) 5679 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5680{ 5681} 5682 5683AudioFlinger::NotificationClient::~NotificationClient() 5684{ 5685} 5686 5687void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5688{ 5689 sp<NotificationClient> keep(this); 5690 mAudioFlinger->removeNotificationClient(mPid); 5691} 5692 5693// ---------------------------------------------------------------------------- 5694 5695AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5696 : BnAudioTrack(), 5697 mTrack(track) 5698{ 5699} 5700 5701AudioFlinger::TrackHandle::~TrackHandle() { 5702 // just stop the track on deletion, associated resources 5703 // will be freed from the main thread once all pending buffers have 5704 // been played. Unless it's not in the active track list, in which 5705 // case we free everything now... 5706 mTrack->destroy(); 5707} 5708 5709sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5710 return mTrack->getCblk(); 5711} 5712 5713status_t AudioFlinger::TrackHandle::start() { 5714 return mTrack->start(); 5715} 5716 5717void AudioFlinger::TrackHandle::stop() { 5718 mTrack->stop(); 5719} 5720 5721void AudioFlinger::TrackHandle::flush() { 5722 mTrack->flush(); 5723} 5724 5725void AudioFlinger::TrackHandle::mute(bool e) { 5726 mTrack->mute(e); 5727} 5728 5729void AudioFlinger::TrackHandle::pause() { 5730 mTrack->pause(); 5731} 5732 5733status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5734{ 5735 return mTrack->attachAuxEffect(EffectId); 5736} 5737 5738status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5739 sp<IMemory>* buffer) { 5740 if (!mTrack->isTimedTrack()) 5741 return INVALID_OPERATION; 5742 5743 PlaybackThread::TimedTrack* tt = 5744 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5745 return tt->allocateTimedBuffer(size, buffer); 5746} 5747 5748status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5749 int64_t pts) { 5750 if (!mTrack->isTimedTrack()) 5751 return INVALID_OPERATION; 5752 5753 PlaybackThread::TimedTrack* tt = 5754 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5755 return tt->queueTimedBuffer(buffer, pts); 5756} 5757 5758status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5759 const LinearTransform& xform, int target) { 5760 5761 if (!mTrack->isTimedTrack()) 5762 return INVALID_OPERATION; 5763 5764 PlaybackThread::TimedTrack* tt = 5765 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5766 return tt->setMediaTimeTransform( 5767 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5768} 5769 5770status_t AudioFlinger::TrackHandle::onTransact( 5771 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5772{ 5773 return BnAudioTrack::onTransact(code, data, reply, flags); 5774} 5775 5776// ---------------------------------------------------------------------------- 5777 5778sp<IAudioRecord> AudioFlinger::openRecord( 5779 pid_t pid, 5780 audio_io_handle_t input, 5781 uint32_t sampleRate, 5782 audio_format_t format, 5783 uint32_t channelMask, 5784 int frameCount, 5785 IAudioFlinger::track_flags_t flags, 5786 int *sessionId, 5787 status_t *status) 5788{ 5789 sp<RecordThread::RecordTrack> recordTrack; 5790 sp<RecordHandle> recordHandle; 5791 sp<Client> client; 5792 status_t lStatus; 5793 RecordThread *thread; 5794 size_t inFrameCount; 5795 int lSessionId; 5796 5797 // check calling permissions 5798 if (!recordingAllowed()) { 5799 lStatus = PERMISSION_DENIED; 5800 goto Exit; 5801 } 5802 5803 // add client to list 5804 { // scope for mLock 5805 Mutex::Autolock _l(mLock); 5806 thread = checkRecordThread_l(input); 5807 if (thread == NULL) { 5808 lStatus = BAD_VALUE; 5809 goto Exit; 5810 } 5811 5812 client = registerPid_l(pid); 5813 5814 // If no audio session id is provided, create one here 5815 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5816 lSessionId = *sessionId; 5817 } else { 5818 lSessionId = nextUniqueId(); 5819 if (sessionId != NULL) { 5820 *sessionId = lSessionId; 5821 } 5822 } 5823 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5824 recordTrack = thread->createRecordTrack_l(client, 5825 sampleRate, 5826 format, 5827 channelMask, 5828 frameCount, 5829 lSessionId, 5830 &lStatus); 5831 } 5832 if (lStatus != NO_ERROR) { 5833 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5834 // destructor is called by the TrackBase destructor with mLock held 5835 client.clear(); 5836 recordTrack.clear(); 5837 goto Exit; 5838 } 5839 5840 // return to handle to client 5841 recordHandle = new RecordHandle(recordTrack); 5842 lStatus = NO_ERROR; 5843 5844Exit: 5845 if (status) { 5846 *status = lStatus; 5847 } 5848 return recordHandle; 5849} 5850 5851// ---------------------------------------------------------------------------- 5852 5853AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5854 : BnAudioRecord(), 5855 mRecordTrack(recordTrack) 5856{ 5857} 5858 5859AudioFlinger::RecordHandle::~RecordHandle() { 5860 stop(); 5861} 5862 5863sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5864 return mRecordTrack->getCblk(); 5865} 5866 5867status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5868 ALOGV("RecordHandle::start()"); 5869 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5870} 5871 5872void AudioFlinger::RecordHandle::stop() { 5873 ALOGV("RecordHandle::stop()"); 5874 mRecordTrack->stop(); 5875} 5876 5877status_t AudioFlinger::RecordHandle::onTransact( 5878 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5879{ 5880 return BnAudioRecord::onTransact(code, data, reply, flags); 5881} 5882 5883// ---------------------------------------------------------------------------- 5884 5885AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5886 AudioStreamIn *input, 5887 uint32_t sampleRate, 5888 uint32_t channels, 5889 audio_io_handle_t id, 5890 uint32_t device) : 5891 ThreadBase(audioFlinger, id, device, RECORD), 5892 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5893 // mRsmpInIndex and mInputBytes set by readInputParameters() 5894 mReqChannelCount(popcount(channels)), 5895 mReqSampleRate(sampleRate) 5896 // mBytesRead is only meaningful while active, and so is cleared in start() 5897 // (but might be better to also clear here for dump?) 5898{ 5899 snprintf(mName, kNameLength, "AudioIn_%X", id); 5900 5901 readInputParameters(); 5902} 5903 5904 5905AudioFlinger::RecordThread::~RecordThread() 5906{ 5907 delete[] mRsmpInBuffer; 5908 delete mResampler; 5909 delete[] mRsmpOutBuffer; 5910} 5911 5912void AudioFlinger::RecordThread::onFirstRef() 5913{ 5914 run(mName, PRIORITY_URGENT_AUDIO); 5915} 5916 5917status_t AudioFlinger::RecordThread::readyToRun() 5918{ 5919 status_t status = initCheck(); 5920 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5921 return status; 5922} 5923 5924bool AudioFlinger::RecordThread::threadLoop() 5925{ 5926 AudioBufferProvider::Buffer buffer; 5927 sp<RecordTrack> activeTrack; 5928 Vector< sp<EffectChain> > effectChains; 5929 5930 nsecs_t lastWarning = 0; 5931 5932 acquireWakeLock(); 5933 5934 // start recording 5935 while (!exitPending()) { 5936 5937 processConfigEvents(); 5938 5939 { // scope for mLock 5940 Mutex::Autolock _l(mLock); 5941 checkForNewParameters_l(); 5942 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5943 if (!mStandby) { 5944 mInput->stream->common.standby(&mInput->stream->common); 5945 mStandby = true; 5946 } 5947 5948 if (exitPending()) break; 5949 5950 releaseWakeLock_l(); 5951 ALOGV("RecordThread: loop stopping"); 5952 // go to sleep 5953 mWaitWorkCV.wait(mLock); 5954 ALOGV("RecordThread: loop starting"); 5955 acquireWakeLock_l(); 5956 continue; 5957 } 5958 if (mActiveTrack != 0) { 5959 if (mActiveTrack->mState == TrackBase::PAUSING) { 5960 if (!mStandby) { 5961 mInput->stream->common.standby(&mInput->stream->common); 5962 mStandby = true; 5963 } 5964 mActiveTrack.clear(); 5965 mStartStopCond.broadcast(); 5966 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5967 if (mReqChannelCount != mActiveTrack->channelCount()) { 5968 mActiveTrack.clear(); 5969 mStartStopCond.broadcast(); 5970 } else if (mBytesRead != 0) { 5971 // record start succeeds only if first read from audio input 5972 // succeeds 5973 if (mBytesRead > 0) { 5974 mActiveTrack->mState = TrackBase::ACTIVE; 5975 } else { 5976 mActiveTrack.clear(); 5977 } 5978 mStartStopCond.broadcast(); 5979 } 5980 mStandby = false; 5981 } 5982 } 5983 lockEffectChains_l(effectChains); 5984 } 5985 5986 if (mActiveTrack != 0) { 5987 if (mActiveTrack->mState != TrackBase::ACTIVE && 5988 mActiveTrack->mState != TrackBase::RESUMING) { 5989 unlockEffectChains(effectChains); 5990 usleep(kRecordThreadSleepUs); 5991 continue; 5992 } 5993 for (size_t i = 0; i < effectChains.size(); i ++) { 5994 effectChains[i]->process_l(); 5995 } 5996 5997 buffer.frameCount = mFrameCount; 5998 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5999 size_t framesOut = buffer.frameCount; 6000 if (mResampler == NULL) { 6001 // no resampling 6002 while (framesOut) { 6003 size_t framesIn = mFrameCount - mRsmpInIndex; 6004 if (framesIn) { 6005 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6006 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6007 if (framesIn > framesOut) 6008 framesIn = framesOut; 6009 mRsmpInIndex += framesIn; 6010 framesOut -= framesIn; 6011 if ((int)mChannelCount == mReqChannelCount || 6012 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6013 memcpy(dst, src, framesIn * mFrameSize); 6014 } else { 6015 int16_t *src16 = (int16_t *)src; 6016 int16_t *dst16 = (int16_t *)dst; 6017 if (mChannelCount == 1) { 6018 while (framesIn--) { 6019 *dst16++ = *src16; 6020 *dst16++ = *src16++; 6021 } 6022 } else { 6023 while (framesIn--) { 6024 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6025 src16 += 2; 6026 } 6027 } 6028 } 6029 } 6030 if (framesOut && mFrameCount == mRsmpInIndex) { 6031 if (framesOut == mFrameCount && 6032 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6033 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6034 framesOut = 0; 6035 } else { 6036 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6037 mRsmpInIndex = 0; 6038 } 6039 if (mBytesRead < 0) { 6040 ALOGE("Error reading audio input"); 6041 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6042 // Force input into standby so that it tries to 6043 // recover at next read attempt 6044 mInput->stream->common.standby(&mInput->stream->common); 6045 usleep(kRecordThreadSleepUs); 6046 } 6047 mRsmpInIndex = mFrameCount; 6048 framesOut = 0; 6049 buffer.frameCount = 0; 6050 } 6051 } 6052 } 6053 } else { 6054 // resampling 6055 6056 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6057 // alter output frame count as if we were expecting stereo samples 6058 if (mChannelCount == 1 && mReqChannelCount == 1) { 6059 framesOut >>= 1; 6060 } 6061 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6062 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6063 // are 32 bit aligned which should be always true. 6064 if (mChannelCount == 2 && mReqChannelCount == 1) { 6065 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6066 // the resampler always outputs stereo samples: do post stereo to mono conversion 6067 int16_t *src = (int16_t *)mRsmpOutBuffer; 6068 int16_t *dst = buffer.i16; 6069 while (framesOut--) { 6070 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6071 src += 2; 6072 } 6073 } else { 6074 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6075 } 6076 6077 } 6078 if (mFramestoDrop == 0) { 6079 mActiveTrack->releaseBuffer(&buffer); 6080 } else { 6081 if (mFramestoDrop > 0) { 6082 mFramestoDrop -= buffer.frameCount; 6083 if (mFramestoDrop <= 0) { 6084 clearSyncStartEvent(); 6085 } 6086 } else { 6087 mFramestoDrop += buffer.frameCount; 6088 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6089 mSyncStartEvent->isCancelled()) { 6090 ALOGW("Synced record %s, session %d, trigger session %d", 6091 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6092 mActiveTrack->sessionId(), 6093 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6094 clearSyncStartEvent(); 6095 } 6096 } 6097 } 6098 mActiveTrack->overflow(); 6099 } 6100 // client isn't retrieving buffers fast enough 6101 else { 6102 if (!mActiveTrack->setOverflow()) { 6103 nsecs_t now = systemTime(); 6104 if ((now - lastWarning) > kWarningThrottleNs) { 6105 ALOGW("RecordThread: buffer overflow"); 6106 lastWarning = now; 6107 } 6108 } 6109 // Release the processor for a while before asking for a new buffer. 6110 // This will give the application more chance to read from the buffer and 6111 // clear the overflow. 6112 usleep(kRecordThreadSleepUs); 6113 } 6114 } 6115 // enable changes in effect chain 6116 unlockEffectChains(effectChains); 6117 effectChains.clear(); 6118 } 6119 6120 if (!mStandby) { 6121 mInput->stream->common.standby(&mInput->stream->common); 6122 } 6123 mActiveTrack.clear(); 6124 6125 mStartStopCond.broadcast(); 6126 6127 releaseWakeLock(); 6128 6129 ALOGV("RecordThread %p exiting", this); 6130 return false; 6131} 6132 6133 6134sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6135 const sp<AudioFlinger::Client>& client, 6136 uint32_t sampleRate, 6137 audio_format_t format, 6138 int channelMask, 6139 int frameCount, 6140 int sessionId, 6141 status_t *status) 6142{ 6143 sp<RecordTrack> track; 6144 status_t lStatus; 6145 6146 lStatus = initCheck(); 6147 if (lStatus != NO_ERROR) { 6148 ALOGE("Audio driver not initialized."); 6149 goto Exit; 6150 } 6151 6152 { // scope for mLock 6153 Mutex::Autolock _l(mLock); 6154 6155 track = new RecordTrack(this, client, sampleRate, 6156 format, channelMask, frameCount, sessionId); 6157 6158 if (track->getCblk() == 0) { 6159 lStatus = NO_MEMORY; 6160 goto Exit; 6161 } 6162 6163 mTrack = track.get(); 6164 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6165 bool suspend = audio_is_bluetooth_sco_device( 6166 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6167 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6168 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6169 } 6170 lStatus = NO_ERROR; 6171 6172Exit: 6173 if (status) { 6174 *status = lStatus; 6175 } 6176 return track; 6177} 6178 6179status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6180 AudioSystem::sync_event_t event, 6181 int triggerSession) 6182{ 6183 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6184 sp<ThreadBase> strongMe = this; 6185 status_t status = NO_ERROR; 6186 6187 if (event == AudioSystem::SYNC_EVENT_NONE) { 6188 clearSyncStartEvent(); 6189 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6190 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6191 triggerSession, 6192 recordTrack->sessionId(), 6193 syncStartEventCallback, 6194 this); 6195 // Sync event can be cancelled by the trigger session if the track is not in a 6196 // compatible state in which case we start record immediately 6197 if (mSyncStartEvent->isCancelled()) { 6198 clearSyncStartEvent(); 6199 } else { 6200 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6201 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6202 } 6203 } 6204 6205 { 6206 AutoMutex lock(mLock); 6207 if (mActiveTrack != 0) { 6208 if (recordTrack != mActiveTrack.get()) { 6209 status = -EBUSY; 6210 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6211 mActiveTrack->mState = TrackBase::ACTIVE; 6212 } 6213 return status; 6214 } 6215 6216 recordTrack->mState = TrackBase::IDLE; 6217 mActiveTrack = recordTrack; 6218 mLock.unlock(); 6219 status_t status = AudioSystem::startInput(mId); 6220 mLock.lock(); 6221 if (status != NO_ERROR) { 6222 mActiveTrack.clear(); 6223 clearSyncStartEvent(); 6224 return status; 6225 } 6226 mRsmpInIndex = mFrameCount; 6227 mBytesRead = 0; 6228 if (mResampler != NULL) { 6229 mResampler->reset(); 6230 } 6231 mActiveTrack->mState = TrackBase::RESUMING; 6232 // signal thread to start 6233 ALOGV("Signal record thread"); 6234 mWaitWorkCV.signal(); 6235 // do not wait for mStartStopCond if exiting 6236 if (exitPending()) { 6237 mActiveTrack.clear(); 6238 status = INVALID_OPERATION; 6239 goto startError; 6240 } 6241 mStartStopCond.wait(mLock); 6242 if (mActiveTrack == 0) { 6243 ALOGV("Record failed to start"); 6244 status = BAD_VALUE; 6245 goto startError; 6246 } 6247 ALOGV("Record started OK"); 6248 return status; 6249 } 6250startError: 6251 AudioSystem::stopInput(mId); 6252 clearSyncStartEvent(); 6253 return status; 6254} 6255 6256void AudioFlinger::RecordThread::clearSyncStartEvent() 6257{ 6258 if (mSyncStartEvent != 0) { 6259 mSyncStartEvent->cancel(); 6260 } 6261 mSyncStartEvent.clear(); 6262 mFramestoDrop = 0; 6263} 6264 6265void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6266{ 6267 sp<SyncEvent> strongEvent = event.promote(); 6268 6269 if (strongEvent != 0) { 6270 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6271 me->handleSyncStartEvent(strongEvent); 6272 } 6273} 6274 6275void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6276{ 6277 if (event == mSyncStartEvent) { 6278 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6279 // from audio HAL 6280 mFramestoDrop = mFrameCount * 2; 6281 } 6282} 6283 6284void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6285 ALOGV("RecordThread::stop"); 6286 sp<ThreadBase> strongMe = this; 6287 { 6288 AutoMutex lock(mLock); 6289 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6290 mActiveTrack->mState = TrackBase::PAUSING; 6291 // do not wait for mStartStopCond if exiting 6292 if (exitPending()) { 6293 return; 6294 } 6295 mStartStopCond.wait(mLock); 6296 // if we have been restarted, recordTrack == mActiveTrack.get() here 6297 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6298 mLock.unlock(); 6299 AudioSystem::stopInput(mId); 6300 mLock.lock(); 6301 ALOGV("Record stopped OK"); 6302 } 6303 } 6304 } 6305} 6306 6307bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6308{ 6309 return false; 6310} 6311 6312status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6313{ 6314 if (!isValidSyncEvent(event)) { 6315 return BAD_VALUE; 6316 } 6317 6318 Mutex::Autolock _l(mLock); 6319 6320 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6321 mTrack->setSyncEvent(event); 6322 return NO_ERROR; 6323 } 6324 return NAME_NOT_FOUND; 6325} 6326 6327status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6328{ 6329 const size_t SIZE = 256; 6330 char buffer[SIZE]; 6331 String8 result; 6332 6333 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6334 result.append(buffer); 6335 6336 if (mActiveTrack != 0) { 6337 result.append("Active Track:\n"); 6338 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6339 mActiveTrack->dump(buffer, SIZE); 6340 result.append(buffer); 6341 6342 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6343 result.append(buffer); 6344 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6345 result.append(buffer); 6346 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6347 result.append(buffer); 6348 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6349 result.append(buffer); 6350 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6351 result.append(buffer); 6352 6353 6354 } else { 6355 result.append("No record client\n"); 6356 } 6357 write(fd, result.string(), result.size()); 6358 6359 dumpBase(fd, args); 6360 dumpEffectChains(fd, args); 6361 6362 return NO_ERROR; 6363} 6364 6365// AudioBufferProvider interface 6366status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6367{ 6368 size_t framesReq = buffer->frameCount; 6369 size_t framesReady = mFrameCount - mRsmpInIndex; 6370 int channelCount; 6371 6372 if (framesReady == 0) { 6373 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6374 if (mBytesRead < 0) { 6375 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6376 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6377 // Force input into standby so that it tries to 6378 // recover at next read attempt 6379 mInput->stream->common.standby(&mInput->stream->common); 6380 usleep(kRecordThreadSleepUs); 6381 } 6382 buffer->raw = NULL; 6383 buffer->frameCount = 0; 6384 return NOT_ENOUGH_DATA; 6385 } 6386 mRsmpInIndex = 0; 6387 framesReady = mFrameCount; 6388 } 6389 6390 if (framesReq > framesReady) { 6391 framesReq = framesReady; 6392 } 6393 6394 if (mChannelCount == 1 && mReqChannelCount == 2) { 6395 channelCount = 1; 6396 } else { 6397 channelCount = 2; 6398 } 6399 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6400 buffer->frameCount = framesReq; 6401 return NO_ERROR; 6402} 6403 6404// AudioBufferProvider interface 6405void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6406{ 6407 mRsmpInIndex += buffer->frameCount; 6408 buffer->frameCount = 0; 6409} 6410 6411bool AudioFlinger::RecordThread::checkForNewParameters_l() 6412{ 6413 bool reconfig = false; 6414 6415 while (!mNewParameters.isEmpty()) { 6416 status_t status = NO_ERROR; 6417 String8 keyValuePair = mNewParameters[0]; 6418 AudioParameter param = AudioParameter(keyValuePair); 6419 int value; 6420 audio_format_t reqFormat = mFormat; 6421 int reqSamplingRate = mReqSampleRate; 6422 int reqChannelCount = mReqChannelCount; 6423 6424 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6425 reqSamplingRate = value; 6426 reconfig = true; 6427 } 6428 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6429 reqFormat = (audio_format_t) value; 6430 reconfig = true; 6431 } 6432 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6433 reqChannelCount = popcount(value); 6434 reconfig = true; 6435 } 6436 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6437 // do not accept frame count changes if tracks are open as the track buffer 6438 // size depends on frame count and correct behavior would not be guaranteed 6439 // if frame count is changed after track creation 6440 if (mActiveTrack != 0) { 6441 status = INVALID_OPERATION; 6442 } else { 6443 reconfig = true; 6444 } 6445 } 6446 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6447 // forward device change to effects that have requested to be 6448 // aware of attached audio device. 6449 for (size_t i = 0; i < mEffectChains.size(); i++) { 6450 mEffectChains[i]->setDevice_l(value); 6451 } 6452 // store input device and output device but do not forward output device to audio HAL. 6453 // Note that status is ignored by the caller for output device 6454 // (see AudioFlinger::setParameters() 6455 if (value & AUDIO_DEVICE_OUT_ALL) { 6456 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6457 status = BAD_VALUE; 6458 } else { 6459 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6460 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6461 if (mTrack != NULL) { 6462 bool suspend = audio_is_bluetooth_sco_device( 6463 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6464 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6465 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6466 } 6467 } 6468 mDevice |= (uint32_t)value; 6469 } 6470 if (status == NO_ERROR) { 6471 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6472 if (status == INVALID_OPERATION) { 6473 mInput->stream->common.standby(&mInput->stream->common); 6474 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6475 keyValuePair.string()); 6476 } 6477 if (reconfig) { 6478 if (status == BAD_VALUE && 6479 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6480 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6481 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6482 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6483 (reqChannelCount <= FCC_2)) { 6484 status = NO_ERROR; 6485 } 6486 if (status == NO_ERROR) { 6487 readInputParameters(); 6488 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6489 } 6490 } 6491 } 6492 6493 mNewParameters.removeAt(0); 6494 6495 mParamStatus = status; 6496 mParamCond.signal(); 6497 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6498 // already timed out waiting for the status and will never signal the condition. 6499 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6500 } 6501 return reconfig; 6502} 6503 6504String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6505{ 6506 char *s; 6507 String8 out_s8 = String8(); 6508 6509 Mutex::Autolock _l(mLock); 6510 if (initCheck() != NO_ERROR) { 6511 return out_s8; 6512 } 6513 6514 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6515 out_s8 = String8(s); 6516 free(s); 6517 return out_s8; 6518} 6519 6520void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6521 AudioSystem::OutputDescriptor desc; 6522 void *param2 = NULL; 6523 6524 switch (event) { 6525 case AudioSystem::INPUT_OPENED: 6526 case AudioSystem::INPUT_CONFIG_CHANGED: 6527 desc.channels = mChannelMask; 6528 desc.samplingRate = mSampleRate; 6529 desc.format = mFormat; 6530 desc.frameCount = mFrameCount; 6531 desc.latency = 0; 6532 param2 = &desc; 6533 break; 6534 6535 case AudioSystem::INPUT_CLOSED: 6536 default: 6537 break; 6538 } 6539 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6540} 6541 6542void AudioFlinger::RecordThread::readInputParameters() 6543{ 6544 delete mRsmpInBuffer; 6545 // mRsmpInBuffer is always assigned a new[] below 6546 delete mRsmpOutBuffer; 6547 mRsmpOutBuffer = NULL; 6548 delete mResampler; 6549 mResampler = NULL; 6550 6551 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6552 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6553 mChannelCount = (uint16_t)popcount(mChannelMask); 6554 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6555 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6556 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6557 mFrameCount = mInputBytes / mFrameSize; 6558 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6559 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6560 6561 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6562 { 6563 int channelCount; 6564 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6565 // stereo to mono post process as the resampler always outputs stereo. 6566 if (mChannelCount == 1 && mReqChannelCount == 2) { 6567 channelCount = 1; 6568 } else { 6569 channelCount = 2; 6570 } 6571 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6572 mResampler->setSampleRate(mSampleRate); 6573 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6574 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6575 6576 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6577 if (mChannelCount == 1 && mReqChannelCount == 1) { 6578 mFrameCount >>= 1; 6579 } 6580 6581 } 6582 mRsmpInIndex = mFrameCount; 6583} 6584 6585unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6586{ 6587 Mutex::Autolock _l(mLock); 6588 if (initCheck() != NO_ERROR) { 6589 return 0; 6590 } 6591 6592 return mInput->stream->get_input_frames_lost(mInput->stream); 6593} 6594 6595uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6596{ 6597 Mutex::Autolock _l(mLock); 6598 uint32_t result = 0; 6599 if (getEffectChain_l(sessionId) != 0) { 6600 result = EFFECT_SESSION; 6601 } 6602 6603 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6604 result |= TRACK_SESSION; 6605 } 6606 6607 return result; 6608} 6609 6610AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6611{ 6612 Mutex::Autolock _l(mLock); 6613 return mTrack; 6614} 6615 6616AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6617{ 6618 Mutex::Autolock _l(mLock); 6619 return mInput; 6620} 6621 6622AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6623{ 6624 Mutex::Autolock _l(mLock); 6625 AudioStreamIn *input = mInput; 6626 mInput = NULL; 6627 return input; 6628} 6629 6630// this method must always be called either with ThreadBase mLock held or inside the thread loop 6631audio_stream_t* AudioFlinger::RecordThread::stream() const 6632{ 6633 if (mInput == NULL) { 6634 return NULL; 6635 } 6636 return &mInput->stream->common; 6637} 6638 6639 6640// ---------------------------------------------------------------------------- 6641 6642audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6643{ 6644 if (!settingsAllowed()) { 6645 return 0; 6646 } 6647 Mutex::Autolock _l(mLock); 6648 return loadHwModule_l(name); 6649} 6650 6651// loadHwModule_l() must be called with AudioFlinger::mLock held 6652audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6653{ 6654 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6655 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6656 ALOGW("loadHwModule() module %s already loaded", name); 6657 return mAudioHwDevs.keyAt(i); 6658 } 6659 } 6660 6661 audio_hw_device_t *dev; 6662 6663 int rc = load_audio_interface(name, &dev); 6664 if (rc) { 6665 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6666 return 0; 6667 } 6668 6669 mHardwareStatus = AUDIO_HW_INIT; 6670 rc = dev->init_check(dev); 6671 mHardwareStatus = AUDIO_HW_IDLE; 6672 if (rc) { 6673 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6674 return 0; 6675 } 6676 6677 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6678 (NULL != dev->set_master_volume)) { 6679 AutoMutex lock(mHardwareLock); 6680 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6681 dev->set_master_volume(dev, mMasterVolume); 6682 mHardwareStatus = AUDIO_HW_IDLE; 6683 } 6684 6685 audio_module_handle_t handle = nextUniqueId(); 6686 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6687 6688 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6689 name, dev->common.module->name, dev->common.module->id, handle); 6690 6691 return handle; 6692 6693} 6694 6695audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6696 audio_devices_t *pDevices, 6697 uint32_t *pSamplingRate, 6698 audio_format_t *pFormat, 6699 audio_channel_mask_t *pChannelMask, 6700 uint32_t *pLatencyMs, 6701 audio_output_flags_t flags) 6702{ 6703 status_t status; 6704 PlaybackThread *thread = NULL; 6705 struct audio_config config = { 6706 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6707 channel_mask: pChannelMask ? *pChannelMask : 0, 6708 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6709 }; 6710 audio_stream_out_t *outStream = NULL; 6711 audio_hw_device_t *outHwDev; 6712 6713 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6714 module, 6715 (pDevices != NULL) ? (int)*pDevices : 0, 6716 config.sample_rate, 6717 config.format, 6718 config.channel_mask, 6719 flags); 6720 6721 if (pDevices == NULL || *pDevices == 0) { 6722 return 0; 6723 } 6724 6725 Mutex::Autolock _l(mLock); 6726 6727 outHwDev = findSuitableHwDev_l(module, *pDevices); 6728 if (outHwDev == NULL) 6729 return 0; 6730 6731 audio_io_handle_t id = nextUniqueId(); 6732 6733 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6734 6735 status = outHwDev->open_output_stream(outHwDev, 6736 id, 6737 *pDevices, 6738 (audio_output_flags_t)flags, 6739 &config, 6740 &outStream); 6741 6742 mHardwareStatus = AUDIO_HW_IDLE; 6743 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6744 outStream, 6745 config.sample_rate, 6746 config.format, 6747 config.channel_mask, 6748 status); 6749 6750 if (status == NO_ERROR && outStream != NULL) { 6751 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6752 6753 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6754 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6755 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6756 thread = new DirectOutputThread(this, output, id, *pDevices); 6757 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6758 } else { 6759 thread = new MixerThread(this, output, id, *pDevices); 6760 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6761 } 6762 mPlaybackThreads.add(id, thread); 6763 6764 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6765 if (pFormat != NULL) *pFormat = config.format; 6766 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6767 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6768 6769 // notify client processes of the new output creation 6770 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6771 6772 // the first primary output opened designates the primary hw device 6773 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6774 ALOGI("Using module %d has the primary audio interface", module); 6775 mPrimaryHardwareDev = outHwDev; 6776 6777 AutoMutex lock(mHardwareLock); 6778 mHardwareStatus = AUDIO_HW_SET_MODE; 6779 outHwDev->set_mode(outHwDev, mMode); 6780 6781 // Determine the level of master volume support the primary audio HAL has, 6782 // and set the initial master volume at the same time. 6783 float initialVolume = 1.0; 6784 mMasterVolumeSupportLvl = MVS_NONE; 6785 6786 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6787 if ((NULL != outHwDev->get_master_volume) && 6788 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6789 mMasterVolumeSupportLvl = MVS_FULL; 6790 } else { 6791 mMasterVolumeSupportLvl = MVS_SETONLY; 6792 initialVolume = 1.0; 6793 } 6794 6795 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6796 if ((NULL == outHwDev->set_master_volume) || 6797 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6798 mMasterVolumeSupportLvl = MVS_NONE; 6799 } 6800 // now that we have a primary device, initialize master volume on other devices 6801 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6802 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6803 6804 if ((dev != mPrimaryHardwareDev) && 6805 (NULL != dev->set_master_volume)) { 6806 dev->set_master_volume(dev, initialVolume); 6807 } 6808 } 6809 mHardwareStatus = AUDIO_HW_IDLE; 6810 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6811 ? initialVolume 6812 : 1.0; 6813 mMasterVolume = initialVolume; 6814 } 6815 return id; 6816 } 6817 6818 return 0; 6819} 6820 6821audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6822 audio_io_handle_t output2) 6823{ 6824 Mutex::Autolock _l(mLock); 6825 MixerThread *thread1 = checkMixerThread_l(output1); 6826 MixerThread *thread2 = checkMixerThread_l(output2); 6827 6828 if (thread1 == NULL || thread2 == NULL) { 6829 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6830 return 0; 6831 } 6832 6833 audio_io_handle_t id = nextUniqueId(); 6834 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6835 thread->addOutputTrack(thread2); 6836 mPlaybackThreads.add(id, thread); 6837 // notify client processes of the new output creation 6838 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6839 return id; 6840} 6841 6842status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6843{ 6844 // keep strong reference on the playback thread so that 6845 // it is not destroyed while exit() is executed 6846 sp<PlaybackThread> thread; 6847 { 6848 Mutex::Autolock _l(mLock); 6849 thread = checkPlaybackThread_l(output); 6850 if (thread == NULL) { 6851 return BAD_VALUE; 6852 } 6853 6854 ALOGV("closeOutput() %d", output); 6855 6856 if (thread->type() == ThreadBase::MIXER) { 6857 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6858 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6859 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6860 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6861 } 6862 } 6863 } 6864 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6865 mPlaybackThreads.removeItem(output); 6866 } 6867 thread->exit(); 6868 // The thread entity (active unit of execution) is no longer running here, 6869 // but the ThreadBase container still exists. 6870 6871 if (thread->type() != ThreadBase::DUPLICATING) { 6872 AudioStreamOut *out = thread->clearOutput(); 6873 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6874 // from now on thread->mOutput is NULL 6875 out->hwDev->close_output_stream(out->hwDev, out->stream); 6876 delete out; 6877 } 6878 return NO_ERROR; 6879} 6880 6881status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6882{ 6883 Mutex::Autolock _l(mLock); 6884 PlaybackThread *thread = checkPlaybackThread_l(output); 6885 6886 if (thread == NULL) { 6887 return BAD_VALUE; 6888 } 6889 6890 ALOGV("suspendOutput() %d", output); 6891 thread->suspend(); 6892 6893 return NO_ERROR; 6894} 6895 6896status_t AudioFlinger::restoreOutput(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("restoreOutput() %d", output); 6906 6907 thread->restore(); 6908 6909 return NO_ERROR; 6910} 6911 6912audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6913 audio_devices_t *pDevices, 6914 uint32_t *pSamplingRate, 6915 audio_format_t *pFormat, 6916 uint32_t *pChannelMask) 6917{ 6918 status_t status; 6919 RecordThread *thread = NULL; 6920 struct audio_config config = { 6921 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6922 channel_mask: pChannelMask ? *pChannelMask : 0, 6923 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6924 }; 6925 uint32_t reqSamplingRate = config.sample_rate; 6926 audio_format_t reqFormat = config.format; 6927 audio_channel_mask_t reqChannels = config.channel_mask; 6928 audio_stream_in_t *inStream = NULL; 6929 audio_hw_device_t *inHwDev; 6930 6931 if (pDevices == NULL || *pDevices == 0) { 6932 return 0; 6933 } 6934 6935 Mutex::Autolock _l(mLock); 6936 6937 inHwDev = findSuitableHwDev_l(module, *pDevices); 6938 if (inHwDev == NULL) 6939 return 0; 6940 6941 audio_io_handle_t id = nextUniqueId(); 6942 6943 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6944 &inStream); 6945 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6946 inStream, 6947 config.sample_rate, 6948 config.format, 6949 config.channel_mask, 6950 status); 6951 6952 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6953 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6954 // or stereo to mono conversions on 16 bit PCM inputs. 6955 if (status == BAD_VALUE && 6956 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6957 (config.sample_rate <= 2 * reqSamplingRate) && 6958 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6959 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6960 inStream = NULL; 6961 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6962 } 6963 6964 if (status == NO_ERROR && inStream != NULL) { 6965 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6966 6967 // Start record thread 6968 // RecorThread require both input and output device indication to forward to audio 6969 // pre processing modules 6970 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6971 thread = new RecordThread(this, 6972 input, 6973 reqSamplingRate, 6974 reqChannels, 6975 id, 6976 device); 6977 mRecordThreads.add(id, thread); 6978 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6979 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6980 if (pFormat != NULL) *pFormat = config.format; 6981 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6982 6983 input->stream->common.standby(&input->stream->common); 6984 6985 // notify client processes of the new input creation 6986 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6987 return id; 6988 } 6989 6990 return 0; 6991} 6992 6993status_t AudioFlinger::closeInput(audio_io_handle_t input) 6994{ 6995 // keep strong reference on the record thread so that 6996 // it is not destroyed while exit() is executed 6997 sp<RecordThread> thread; 6998 { 6999 Mutex::Autolock _l(mLock); 7000 thread = checkRecordThread_l(input); 7001 if (thread == NULL) { 7002 return BAD_VALUE; 7003 } 7004 7005 ALOGV("closeInput() %d", input); 7006 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7007 mRecordThreads.removeItem(input); 7008 } 7009 thread->exit(); 7010 // The thread entity (active unit of execution) is no longer running here, 7011 // but the ThreadBase container still exists. 7012 7013 AudioStreamIn *in = thread->clearInput(); 7014 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7015 // from now on thread->mInput is NULL 7016 in->hwDev->close_input_stream(in->hwDev, in->stream); 7017 delete in; 7018 7019 return NO_ERROR; 7020} 7021 7022status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7023{ 7024 Mutex::Autolock _l(mLock); 7025 MixerThread *dstThread = checkMixerThread_l(output); 7026 if (dstThread == NULL) { 7027 ALOGW("setStreamOutput() bad output id %d", output); 7028 return BAD_VALUE; 7029 } 7030 7031 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7032 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 7033 7034 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7035 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7036 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 7037 MixerThread *srcThread = (MixerThread *)thread; 7038 srcThread->invalidateTracks(stream); 7039 } 7040 } 7041 7042 return NO_ERROR; 7043} 7044 7045 7046int AudioFlinger::newAudioSessionId() 7047{ 7048 return nextUniqueId(); 7049} 7050 7051void AudioFlinger::acquireAudioSessionId(int audioSession) 7052{ 7053 Mutex::Autolock _l(mLock); 7054 pid_t caller = IPCThreadState::self()->getCallingPid(); 7055 ALOGV("acquiring %d from %d", audioSession, caller); 7056 size_t num = mAudioSessionRefs.size(); 7057 for (size_t i = 0; i< num; i++) { 7058 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7059 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7060 ref->mCnt++; 7061 ALOGV(" incremented refcount to %d", ref->mCnt); 7062 return; 7063 } 7064 } 7065 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7066 ALOGV(" added new entry for %d", audioSession); 7067} 7068 7069void AudioFlinger::releaseAudioSessionId(int audioSession) 7070{ 7071 Mutex::Autolock _l(mLock); 7072 pid_t caller = IPCThreadState::self()->getCallingPid(); 7073 ALOGV("releasing %d from %d", audioSession, caller); 7074 size_t num = mAudioSessionRefs.size(); 7075 for (size_t i = 0; i< num; i++) { 7076 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7077 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7078 ref->mCnt--; 7079 ALOGV(" decremented refcount to %d", ref->mCnt); 7080 if (ref->mCnt == 0) { 7081 mAudioSessionRefs.removeAt(i); 7082 delete ref; 7083 purgeStaleEffects_l(); 7084 } 7085 return; 7086 } 7087 } 7088 ALOGW("session id %d not found for pid %d", audioSession, caller); 7089} 7090 7091void AudioFlinger::purgeStaleEffects_l() { 7092 7093 ALOGV("purging stale effects"); 7094 7095 Vector< sp<EffectChain> > chains; 7096 7097 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7098 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7099 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7100 sp<EffectChain> ec = t->mEffectChains[j]; 7101 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7102 chains.push(ec); 7103 } 7104 } 7105 } 7106 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7107 sp<RecordThread> t = mRecordThreads.valueAt(i); 7108 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7109 sp<EffectChain> ec = t->mEffectChains[j]; 7110 chains.push(ec); 7111 } 7112 } 7113 7114 for (size_t i = 0; i < chains.size(); i++) { 7115 sp<EffectChain> ec = chains[i]; 7116 int sessionid = ec->sessionId(); 7117 sp<ThreadBase> t = ec->mThread.promote(); 7118 if (t == 0) { 7119 continue; 7120 } 7121 size_t numsessionrefs = mAudioSessionRefs.size(); 7122 bool found = false; 7123 for (size_t k = 0; k < numsessionrefs; k++) { 7124 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7125 if (ref->mSessionid == sessionid) { 7126 ALOGV(" session %d still exists for %d with %d refs", 7127 sessionid, ref->mPid, ref->mCnt); 7128 found = true; 7129 break; 7130 } 7131 } 7132 if (!found) { 7133 // remove all effects from the chain 7134 while (ec->mEffects.size()) { 7135 sp<EffectModule> effect = ec->mEffects[0]; 7136 effect->unPin(); 7137 Mutex::Autolock _l (t->mLock); 7138 t->removeEffect_l(effect); 7139 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7140 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7141 if (handle != 0) { 7142 handle->mEffect.clear(); 7143 if (handle->mHasControl && handle->mEnabled) { 7144 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7145 } 7146 } 7147 } 7148 AudioSystem::unregisterEffect(effect->id()); 7149 } 7150 } 7151 } 7152 return; 7153} 7154 7155// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7156AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7157{ 7158 return mPlaybackThreads.valueFor(output).get(); 7159} 7160 7161// checkMixerThread_l() must be called with AudioFlinger::mLock held 7162AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7163{ 7164 PlaybackThread *thread = checkPlaybackThread_l(output); 7165 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7166} 7167 7168// checkRecordThread_l() must be called with AudioFlinger::mLock held 7169AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7170{ 7171 return mRecordThreads.valueFor(input).get(); 7172} 7173 7174uint32_t AudioFlinger::nextUniqueId() 7175{ 7176 return android_atomic_inc(&mNextUniqueId); 7177} 7178 7179AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7180{ 7181 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7182 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7183 AudioStreamOut *output = thread->getOutput(); 7184 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7185 return thread; 7186 } 7187 } 7188 return NULL; 7189} 7190 7191uint32_t AudioFlinger::primaryOutputDevice_l() const 7192{ 7193 PlaybackThread *thread = primaryPlaybackThread_l(); 7194 7195 if (thread == NULL) { 7196 return 0; 7197 } 7198 7199 return thread->device(); 7200} 7201 7202sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7203 int triggerSession, 7204 int listenerSession, 7205 sync_event_callback_t callBack, 7206 void *cookie) 7207{ 7208 Mutex::Autolock _l(mLock); 7209 7210 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7211 status_t playStatus = NAME_NOT_FOUND; 7212 status_t recStatus = NAME_NOT_FOUND; 7213 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7214 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7215 if (playStatus == NO_ERROR) { 7216 return event; 7217 } 7218 } 7219 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7220 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7221 if (recStatus == NO_ERROR) { 7222 return event; 7223 } 7224 } 7225 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7226 mPendingSyncEvents.add(event); 7227 } else { 7228 ALOGV("createSyncEvent() invalid event %d", event->type()); 7229 event.clear(); 7230 } 7231 return event; 7232} 7233 7234// ---------------------------------------------------------------------------- 7235// Effect management 7236// ---------------------------------------------------------------------------- 7237 7238 7239status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7240{ 7241 Mutex::Autolock _l(mLock); 7242 return EffectQueryNumberEffects(numEffects); 7243} 7244 7245status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7246{ 7247 Mutex::Autolock _l(mLock); 7248 return EffectQueryEffect(index, descriptor); 7249} 7250 7251status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7252 effect_descriptor_t *descriptor) const 7253{ 7254 Mutex::Autolock _l(mLock); 7255 return EffectGetDescriptor(pUuid, descriptor); 7256} 7257 7258 7259sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7260 effect_descriptor_t *pDesc, 7261 const sp<IEffectClient>& effectClient, 7262 int32_t priority, 7263 audio_io_handle_t io, 7264 int sessionId, 7265 status_t *status, 7266 int *id, 7267 int *enabled) 7268{ 7269 status_t lStatus = NO_ERROR; 7270 sp<EffectHandle> handle; 7271 effect_descriptor_t desc; 7272 7273 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7274 pid, effectClient.get(), priority, sessionId, io); 7275 7276 if (pDesc == NULL) { 7277 lStatus = BAD_VALUE; 7278 goto Exit; 7279 } 7280 7281 // check audio settings permission for global effects 7282 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7283 lStatus = PERMISSION_DENIED; 7284 goto Exit; 7285 } 7286 7287 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7288 // that can only be created by audio policy manager (running in same process) 7289 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7290 lStatus = PERMISSION_DENIED; 7291 goto Exit; 7292 } 7293 7294 if (io == 0) { 7295 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7296 // output must be specified by AudioPolicyManager when using session 7297 // AUDIO_SESSION_OUTPUT_STAGE 7298 lStatus = BAD_VALUE; 7299 goto Exit; 7300 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7301 // if the output returned by getOutputForEffect() is removed before we lock the 7302 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7303 // and we will exit safely 7304 io = AudioSystem::getOutputForEffect(&desc); 7305 } 7306 } 7307 7308 { 7309 Mutex::Autolock _l(mLock); 7310 7311 7312 if (!EffectIsNullUuid(&pDesc->uuid)) { 7313 // if uuid is specified, request effect descriptor 7314 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7315 if (lStatus < 0) { 7316 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7317 goto Exit; 7318 } 7319 } else { 7320 // if uuid is not specified, look for an available implementation 7321 // of the required type in effect factory 7322 if (EffectIsNullUuid(&pDesc->type)) { 7323 ALOGW("createEffect() no effect type"); 7324 lStatus = BAD_VALUE; 7325 goto Exit; 7326 } 7327 uint32_t numEffects = 0; 7328 effect_descriptor_t d; 7329 d.flags = 0; // prevent compiler warning 7330 bool found = false; 7331 7332 lStatus = EffectQueryNumberEffects(&numEffects); 7333 if (lStatus < 0) { 7334 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7335 goto Exit; 7336 } 7337 for (uint32_t i = 0; i < numEffects; i++) { 7338 lStatus = EffectQueryEffect(i, &desc); 7339 if (lStatus < 0) { 7340 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7341 continue; 7342 } 7343 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7344 // If matching type found save effect descriptor. If the session is 7345 // 0 and the effect is not auxiliary, continue enumeration in case 7346 // an auxiliary version of this effect type is available 7347 found = true; 7348 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7349 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7350 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7351 break; 7352 } 7353 } 7354 } 7355 if (!found) { 7356 lStatus = BAD_VALUE; 7357 ALOGW("createEffect() effect not found"); 7358 goto Exit; 7359 } 7360 // For same effect type, chose auxiliary version over insert version if 7361 // connect to output mix (Compliance to OpenSL ES) 7362 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7363 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7364 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7365 } 7366 } 7367 7368 // Do not allow auxiliary effects on a session different from 0 (output mix) 7369 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7370 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7371 lStatus = INVALID_OPERATION; 7372 goto Exit; 7373 } 7374 7375 // check recording permission for visualizer 7376 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7377 !recordingAllowed()) { 7378 lStatus = PERMISSION_DENIED; 7379 goto Exit; 7380 } 7381 7382 // return effect descriptor 7383 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7384 7385 // If output is not specified try to find a matching audio session ID in one of the 7386 // output threads. 7387 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7388 // because of code checking output when entering the function. 7389 // Note: io is never 0 when creating an effect on an input 7390 if (io == 0) { 7391 // look for the thread where the specified audio session is present 7392 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7393 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7394 io = mPlaybackThreads.keyAt(i); 7395 break; 7396 } 7397 } 7398 if (io == 0) { 7399 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7400 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7401 io = mRecordThreads.keyAt(i); 7402 break; 7403 } 7404 } 7405 } 7406 // If no output thread contains the requested session ID, default to 7407 // first output. The effect chain will be moved to the correct output 7408 // thread when a track with the same session ID is created 7409 if (io == 0 && mPlaybackThreads.size()) { 7410 io = mPlaybackThreads.keyAt(0); 7411 } 7412 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7413 } 7414 ThreadBase *thread = checkRecordThread_l(io); 7415 if (thread == NULL) { 7416 thread = checkPlaybackThread_l(io); 7417 if (thread == NULL) { 7418 ALOGE("createEffect() unknown output thread"); 7419 lStatus = BAD_VALUE; 7420 goto Exit; 7421 } 7422 } 7423 7424 sp<Client> client = registerPid_l(pid); 7425 7426 // create effect on selected output thread 7427 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7428 &desc, enabled, &lStatus); 7429 if (handle != 0 && id != NULL) { 7430 *id = handle->id(); 7431 } 7432 } 7433 7434Exit: 7435 if (status != NULL) { 7436 *status = lStatus; 7437 } 7438 return handle; 7439} 7440 7441status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7442 audio_io_handle_t dstOutput) 7443{ 7444 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7445 sessionId, srcOutput, dstOutput); 7446 Mutex::Autolock _l(mLock); 7447 if (srcOutput == dstOutput) { 7448 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7449 return NO_ERROR; 7450 } 7451 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7452 if (srcThread == NULL) { 7453 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7454 return BAD_VALUE; 7455 } 7456 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7457 if (dstThread == NULL) { 7458 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7459 return BAD_VALUE; 7460 } 7461 7462 Mutex::Autolock _dl(dstThread->mLock); 7463 Mutex::Autolock _sl(srcThread->mLock); 7464 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7465 7466 return NO_ERROR; 7467} 7468 7469// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7470status_t AudioFlinger::moveEffectChain_l(int sessionId, 7471 AudioFlinger::PlaybackThread *srcThread, 7472 AudioFlinger::PlaybackThread *dstThread, 7473 bool reRegister) 7474{ 7475 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7476 sessionId, srcThread, dstThread); 7477 7478 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7479 if (chain == 0) { 7480 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7481 sessionId, srcThread); 7482 return INVALID_OPERATION; 7483 } 7484 7485 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7486 // so that a new chain is created with correct parameters when first effect is added. This is 7487 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7488 // removed. 7489 srcThread->removeEffectChain_l(chain); 7490 7491 // transfer all effects one by one so that new effect chain is created on new thread with 7492 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7493 audio_io_handle_t dstOutput = dstThread->id(); 7494 sp<EffectChain> dstChain; 7495 uint32_t strategy = 0; // prevent compiler warning 7496 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7497 while (effect != 0) { 7498 srcThread->removeEffect_l(effect); 7499 dstThread->addEffect_l(effect); 7500 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7501 if (effect->state() == EffectModule::ACTIVE || 7502 effect->state() == EffectModule::STOPPING) { 7503 effect->start(); 7504 } 7505 // if the move request is not received from audio policy manager, the effect must be 7506 // re-registered with the new strategy and output 7507 if (dstChain == 0) { 7508 dstChain = effect->chain().promote(); 7509 if (dstChain == 0) { 7510 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7511 srcThread->addEffect_l(effect); 7512 return NO_INIT; 7513 } 7514 strategy = dstChain->strategy(); 7515 } 7516 if (reRegister) { 7517 AudioSystem::unregisterEffect(effect->id()); 7518 AudioSystem::registerEffect(&effect->desc(), 7519 dstOutput, 7520 strategy, 7521 sessionId, 7522 effect->id()); 7523 } 7524 effect = chain->getEffectFromId_l(0); 7525 } 7526 7527 return NO_ERROR; 7528} 7529 7530 7531// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7532sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7533 const sp<AudioFlinger::Client>& client, 7534 const sp<IEffectClient>& effectClient, 7535 int32_t priority, 7536 int sessionId, 7537 effect_descriptor_t *desc, 7538 int *enabled, 7539 status_t *status 7540 ) 7541{ 7542 sp<EffectModule> effect; 7543 sp<EffectHandle> handle; 7544 status_t lStatus; 7545 sp<EffectChain> chain; 7546 bool chainCreated = false; 7547 bool effectCreated = false; 7548 bool effectRegistered = false; 7549 7550 lStatus = initCheck(); 7551 if (lStatus != NO_ERROR) { 7552 ALOGW("createEffect_l() Audio driver not initialized."); 7553 goto Exit; 7554 } 7555 7556 // Do not allow effects with session ID 0 on direct output or duplicating threads 7557 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7558 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7559 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7560 desc->name, sessionId); 7561 lStatus = BAD_VALUE; 7562 goto Exit; 7563 } 7564 // Only Pre processor effects are allowed on input threads and only on input threads 7565 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7566 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7567 desc->name, desc->flags, mType); 7568 lStatus = BAD_VALUE; 7569 goto Exit; 7570 } 7571 7572 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7573 7574 { // scope for mLock 7575 Mutex::Autolock _l(mLock); 7576 7577 // check for existing effect chain with the requested audio session 7578 chain = getEffectChain_l(sessionId); 7579 if (chain == 0) { 7580 // create a new chain for this session 7581 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7582 chain = new EffectChain(this, sessionId); 7583 addEffectChain_l(chain); 7584 chain->setStrategy(getStrategyForSession_l(sessionId)); 7585 chainCreated = true; 7586 } else { 7587 effect = chain->getEffectFromDesc_l(desc); 7588 } 7589 7590 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7591 7592 if (effect == 0) { 7593 int id = mAudioFlinger->nextUniqueId(); 7594 // Check CPU and memory usage 7595 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7596 if (lStatus != NO_ERROR) { 7597 goto Exit; 7598 } 7599 effectRegistered = true; 7600 // create a new effect module if none present in the chain 7601 effect = new EffectModule(this, chain, desc, id, sessionId); 7602 lStatus = effect->status(); 7603 if (lStatus != NO_ERROR) { 7604 goto Exit; 7605 } 7606 lStatus = chain->addEffect_l(effect); 7607 if (lStatus != NO_ERROR) { 7608 goto Exit; 7609 } 7610 effectCreated = true; 7611 7612 effect->setDevice(mDevice); 7613 effect->setMode(mAudioFlinger->getMode()); 7614 } 7615 // create effect handle and connect it to effect module 7616 handle = new EffectHandle(effect, client, effectClient, priority); 7617 lStatus = effect->addHandle(handle); 7618 if (enabled != NULL) { 7619 *enabled = (int)effect->isEnabled(); 7620 } 7621 } 7622 7623Exit: 7624 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7625 Mutex::Autolock _l(mLock); 7626 if (effectCreated) { 7627 chain->removeEffect_l(effect); 7628 } 7629 if (effectRegistered) { 7630 AudioSystem::unregisterEffect(effect->id()); 7631 } 7632 if (chainCreated) { 7633 removeEffectChain_l(chain); 7634 } 7635 handle.clear(); 7636 } 7637 7638 if (status != NULL) { 7639 *status = lStatus; 7640 } 7641 return handle; 7642} 7643 7644sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7645{ 7646 sp<EffectChain> chain = getEffectChain_l(sessionId); 7647 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7648} 7649 7650// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7651// PlaybackThread::mLock held 7652status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7653{ 7654 // check for existing effect chain with the requested audio session 7655 int sessionId = effect->sessionId(); 7656 sp<EffectChain> chain = getEffectChain_l(sessionId); 7657 bool chainCreated = false; 7658 7659 if (chain == 0) { 7660 // create a new chain for this session 7661 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7662 chain = new EffectChain(this, sessionId); 7663 addEffectChain_l(chain); 7664 chain->setStrategy(getStrategyForSession_l(sessionId)); 7665 chainCreated = true; 7666 } 7667 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7668 7669 if (chain->getEffectFromId_l(effect->id()) != 0) { 7670 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7671 this, effect->desc().name, chain.get()); 7672 return BAD_VALUE; 7673 } 7674 7675 status_t status = chain->addEffect_l(effect); 7676 if (status != NO_ERROR) { 7677 if (chainCreated) { 7678 removeEffectChain_l(chain); 7679 } 7680 return status; 7681 } 7682 7683 effect->setDevice(mDevice); 7684 effect->setMode(mAudioFlinger->getMode()); 7685 return NO_ERROR; 7686} 7687 7688void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7689 7690 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7691 effect_descriptor_t desc = effect->desc(); 7692 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7693 detachAuxEffect_l(effect->id()); 7694 } 7695 7696 sp<EffectChain> chain = effect->chain().promote(); 7697 if (chain != 0) { 7698 // remove effect chain if removing last effect 7699 if (chain->removeEffect_l(effect) == 0) { 7700 removeEffectChain_l(chain); 7701 } 7702 } else { 7703 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7704 } 7705} 7706 7707void AudioFlinger::ThreadBase::lockEffectChains_l( 7708 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7709{ 7710 effectChains = mEffectChains; 7711 for (size_t i = 0; i < mEffectChains.size(); i++) { 7712 mEffectChains[i]->lock(); 7713 } 7714} 7715 7716void AudioFlinger::ThreadBase::unlockEffectChains( 7717 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7718{ 7719 for (size_t i = 0; i < effectChains.size(); i++) { 7720 effectChains[i]->unlock(); 7721 } 7722} 7723 7724sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7725{ 7726 Mutex::Autolock _l(mLock); 7727 return getEffectChain_l(sessionId); 7728} 7729 7730sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7731{ 7732 size_t size = mEffectChains.size(); 7733 for (size_t i = 0; i < size; i++) { 7734 if (mEffectChains[i]->sessionId() == sessionId) { 7735 return mEffectChains[i]; 7736 } 7737 } 7738 return 0; 7739} 7740 7741void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7742{ 7743 Mutex::Autolock _l(mLock); 7744 size_t size = mEffectChains.size(); 7745 for (size_t i = 0; i < size; i++) { 7746 mEffectChains[i]->setMode_l(mode); 7747 } 7748} 7749 7750void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7751 const wp<EffectHandle>& handle, 7752 bool unpinIfLast) { 7753 7754 Mutex::Autolock _l(mLock); 7755 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7756 // delete the effect module if removing last handle on it 7757 if (effect->removeHandle(handle) == 0) { 7758 if (!effect->isPinned() || unpinIfLast) { 7759 removeEffect_l(effect); 7760 AudioSystem::unregisterEffect(effect->id()); 7761 } 7762 } 7763} 7764 7765status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7766{ 7767 int session = chain->sessionId(); 7768 int16_t *buffer = mMixBuffer; 7769 bool ownsBuffer = false; 7770 7771 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7772 if (session > 0) { 7773 // Only one effect chain can be present in direct output thread and it uses 7774 // the mix buffer as input 7775 if (mType != DIRECT) { 7776 size_t numSamples = mNormalFrameCount * mChannelCount; 7777 buffer = new int16_t[numSamples]; 7778 memset(buffer, 0, numSamples * sizeof(int16_t)); 7779 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7780 ownsBuffer = true; 7781 } 7782 7783 // Attach all tracks with same session ID to this chain. 7784 for (size_t i = 0; i < mTracks.size(); ++i) { 7785 sp<Track> track = mTracks[i]; 7786 if (session == track->sessionId()) { 7787 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7788 track->setMainBuffer(buffer); 7789 chain->incTrackCnt(); 7790 } 7791 } 7792 7793 // indicate all active tracks in the chain 7794 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7795 sp<Track> track = mActiveTracks[i].promote(); 7796 if (track == 0) continue; 7797 if (session == track->sessionId()) { 7798 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7799 chain->incActiveTrackCnt(); 7800 } 7801 } 7802 } 7803 7804 chain->setInBuffer(buffer, ownsBuffer); 7805 chain->setOutBuffer(mMixBuffer); 7806 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7807 // chains list in order to be processed last as it contains output stage effects 7808 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7809 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7810 // after track specific effects and before output stage 7811 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7812 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7813 // Effect chain for other sessions are inserted at beginning of effect 7814 // chains list to be processed before output mix effects. Relative order between other 7815 // sessions is not important 7816 size_t size = mEffectChains.size(); 7817 size_t i = 0; 7818 for (i = 0; i < size; i++) { 7819 if (mEffectChains[i]->sessionId() < session) break; 7820 } 7821 mEffectChains.insertAt(chain, i); 7822 checkSuspendOnAddEffectChain_l(chain); 7823 7824 return NO_ERROR; 7825} 7826 7827size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7828{ 7829 int session = chain->sessionId(); 7830 7831 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7832 7833 for (size_t i = 0; i < mEffectChains.size(); i++) { 7834 if (chain == mEffectChains[i]) { 7835 mEffectChains.removeAt(i); 7836 // detach all active tracks from the chain 7837 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7838 sp<Track> track = mActiveTracks[i].promote(); 7839 if (track == 0) continue; 7840 if (session == track->sessionId()) { 7841 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7842 chain.get(), session); 7843 chain->decActiveTrackCnt(); 7844 } 7845 } 7846 7847 // detach all tracks with same session ID from this chain 7848 for (size_t i = 0; i < mTracks.size(); ++i) { 7849 sp<Track> track = mTracks[i]; 7850 if (session == track->sessionId()) { 7851 track->setMainBuffer(mMixBuffer); 7852 chain->decTrackCnt(); 7853 } 7854 } 7855 break; 7856 } 7857 } 7858 return mEffectChains.size(); 7859} 7860 7861status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7862 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7863{ 7864 Mutex::Autolock _l(mLock); 7865 return attachAuxEffect_l(track, EffectId); 7866} 7867 7868status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7869 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7870{ 7871 status_t status = NO_ERROR; 7872 7873 if (EffectId == 0) { 7874 track->setAuxBuffer(0, NULL); 7875 } else { 7876 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7877 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7878 if (effect != 0) { 7879 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7880 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7881 } else { 7882 status = INVALID_OPERATION; 7883 } 7884 } else { 7885 status = BAD_VALUE; 7886 } 7887 } 7888 return status; 7889} 7890 7891void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7892{ 7893 for (size_t i = 0; i < mTracks.size(); ++i) { 7894 sp<Track> track = mTracks[i]; 7895 if (track->auxEffectId() == effectId) { 7896 attachAuxEffect_l(track, 0); 7897 } 7898 } 7899} 7900 7901status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7902{ 7903 // only one chain per input thread 7904 if (mEffectChains.size() != 0) { 7905 return INVALID_OPERATION; 7906 } 7907 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7908 7909 chain->setInBuffer(NULL); 7910 chain->setOutBuffer(NULL); 7911 7912 checkSuspendOnAddEffectChain_l(chain); 7913 7914 mEffectChains.add(chain); 7915 7916 return NO_ERROR; 7917} 7918 7919size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7920{ 7921 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7922 ALOGW_IF(mEffectChains.size() != 1, 7923 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7924 chain.get(), mEffectChains.size(), this); 7925 if (mEffectChains.size() == 1) { 7926 mEffectChains.removeAt(0); 7927 } 7928 return 0; 7929} 7930 7931// ---------------------------------------------------------------------------- 7932// EffectModule implementation 7933// ---------------------------------------------------------------------------- 7934 7935#undef LOG_TAG 7936#define LOG_TAG "AudioFlinger::EffectModule" 7937 7938AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7939 const wp<AudioFlinger::EffectChain>& chain, 7940 effect_descriptor_t *desc, 7941 int id, 7942 int sessionId) 7943 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7944 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7945{ 7946 ALOGV("Constructor %p", this); 7947 int lStatus; 7948 if (thread == NULL) { 7949 return; 7950 } 7951 7952 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7953 7954 // create effect engine from effect factory 7955 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7956 7957 if (mStatus != NO_ERROR) { 7958 return; 7959 } 7960 lStatus = init(); 7961 if (lStatus < 0) { 7962 mStatus = lStatus; 7963 goto Error; 7964 } 7965 7966 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7967 mPinned = true; 7968 } 7969 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7970 return; 7971Error: 7972 EffectRelease(mEffectInterface); 7973 mEffectInterface = NULL; 7974 ALOGV("Constructor Error %d", mStatus); 7975} 7976 7977AudioFlinger::EffectModule::~EffectModule() 7978{ 7979 ALOGV("Destructor %p", this); 7980 if (mEffectInterface != NULL) { 7981 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7982 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7983 sp<ThreadBase> thread = mThread.promote(); 7984 if (thread != 0) { 7985 audio_stream_t *stream = thread->stream(); 7986 if (stream != NULL) { 7987 stream->remove_audio_effect(stream, mEffectInterface); 7988 } 7989 } 7990 } 7991 // release effect engine 7992 EffectRelease(mEffectInterface); 7993 } 7994} 7995 7996status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7997{ 7998 status_t status; 7999 8000 Mutex::Autolock _l(mLock); 8001 int priority = handle->priority(); 8002 size_t size = mHandles.size(); 8003 sp<EffectHandle> h; 8004 size_t i; 8005 for (i = 0; i < size; i++) { 8006 h = mHandles[i].promote(); 8007 if (h == 0) continue; 8008 if (h->priority() <= priority) break; 8009 } 8010 // if inserted in first place, move effect control from previous owner to this handle 8011 if (i == 0) { 8012 bool enabled = false; 8013 if (h != 0) { 8014 enabled = h->enabled(); 8015 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8016 } 8017 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8018 status = NO_ERROR; 8019 } else { 8020 status = ALREADY_EXISTS; 8021 } 8022 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 8023 mHandles.insertAt(handle, i); 8024 return status; 8025} 8026 8027size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 8028{ 8029 Mutex::Autolock _l(mLock); 8030 size_t size = mHandles.size(); 8031 size_t i; 8032 for (i = 0; i < size; i++) { 8033 if (mHandles[i] == handle) break; 8034 } 8035 if (i == size) { 8036 return size; 8037 } 8038 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 8039 8040 bool enabled = false; 8041 EffectHandle *hdl = handle.unsafe_get(); 8042 if (hdl != NULL) { 8043 ALOGV("removeHandle() unsafe_get OK"); 8044 enabled = hdl->enabled(); 8045 } 8046 mHandles.removeAt(i); 8047 size = mHandles.size(); 8048 // if removed from first place, move effect control from this handle to next in line 8049 if (i == 0 && size != 0) { 8050 sp<EffectHandle> h = mHandles[0].promote(); 8051 if (h != 0) { 8052 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 8053 } 8054 } 8055 8056 // Prevent calls to process() and other functions on effect interface from now on. 8057 // The effect engine will be released by the destructor when the last strong reference on 8058 // this object is released which can happen after next process is called. 8059 if (size == 0 && !mPinned) { 8060 mState = DESTROYED; 8061 } 8062 8063 return size; 8064} 8065 8066sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8067{ 8068 Mutex::Autolock _l(mLock); 8069 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8070} 8071 8072void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8073{ 8074 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8075 // keep a strong reference on this EffectModule to avoid calling the 8076 // destructor before we exit 8077 sp<EffectModule> keep(this); 8078 { 8079 sp<ThreadBase> thread = mThread.promote(); 8080 if (thread != 0) { 8081 thread->disconnectEffect(keep, handle, unpinIfLast); 8082 } 8083 } 8084} 8085 8086void AudioFlinger::EffectModule::updateState() { 8087 Mutex::Autolock _l(mLock); 8088 8089 switch (mState) { 8090 case RESTART: 8091 reset_l(); 8092 // FALL THROUGH 8093 8094 case STARTING: 8095 // clear auxiliary effect input buffer for next accumulation 8096 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8097 memset(mConfig.inputCfg.buffer.raw, 8098 0, 8099 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8100 } 8101 start_l(); 8102 mState = ACTIVE; 8103 break; 8104 case STOPPING: 8105 stop_l(); 8106 mDisableWaitCnt = mMaxDisableWaitCnt; 8107 mState = STOPPED; 8108 break; 8109 case STOPPED: 8110 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8111 // turn off sequence. 8112 if (--mDisableWaitCnt == 0) { 8113 reset_l(); 8114 mState = IDLE; 8115 } 8116 break; 8117 default: //IDLE , ACTIVE, DESTROYED 8118 break; 8119 } 8120} 8121 8122void AudioFlinger::EffectModule::process() 8123{ 8124 Mutex::Autolock _l(mLock); 8125 8126 if (mState == DESTROYED || mEffectInterface == NULL || 8127 mConfig.inputCfg.buffer.raw == NULL || 8128 mConfig.outputCfg.buffer.raw == NULL) { 8129 return; 8130 } 8131 8132 if (isProcessEnabled()) { 8133 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8134 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8135 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8136 mConfig.inputCfg.buffer.s32, 8137 mConfig.inputCfg.buffer.frameCount/2); 8138 } 8139 8140 // do the actual processing in the effect engine 8141 int ret = (*mEffectInterface)->process(mEffectInterface, 8142 &mConfig.inputCfg.buffer, 8143 &mConfig.outputCfg.buffer); 8144 8145 // force transition to IDLE state when engine is ready 8146 if (mState == STOPPED && ret == -ENODATA) { 8147 mDisableWaitCnt = 1; 8148 } 8149 8150 // clear auxiliary effect input buffer for next accumulation 8151 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8152 memset(mConfig.inputCfg.buffer.raw, 0, 8153 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8154 } 8155 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8156 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8157 // If an insert effect is idle and input buffer is different from output buffer, 8158 // accumulate input onto output 8159 sp<EffectChain> chain = mChain.promote(); 8160 if (chain != 0 && chain->activeTrackCnt() != 0) { 8161 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8162 int16_t *in = mConfig.inputCfg.buffer.s16; 8163 int16_t *out = mConfig.outputCfg.buffer.s16; 8164 for (size_t i = 0; i < frameCnt; i++) { 8165 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8166 } 8167 } 8168 } 8169} 8170 8171void AudioFlinger::EffectModule::reset_l() 8172{ 8173 if (mEffectInterface == NULL) { 8174 return; 8175 } 8176 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8177} 8178 8179status_t AudioFlinger::EffectModule::configure() 8180{ 8181 uint32_t channels; 8182 if (mEffectInterface == NULL) { 8183 return NO_INIT; 8184 } 8185 8186 sp<ThreadBase> thread = mThread.promote(); 8187 if (thread == 0) { 8188 return DEAD_OBJECT; 8189 } 8190 8191 // TODO: handle configuration of effects replacing track process 8192 if (thread->channelCount() == 1) { 8193 channels = AUDIO_CHANNEL_OUT_MONO; 8194 } else { 8195 channels = AUDIO_CHANNEL_OUT_STEREO; 8196 } 8197 8198 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8199 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8200 } else { 8201 mConfig.inputCfg.channels = channels; 8202 } 8203 mConfig.outputCfg.channels = channels; 8204 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8205 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8206 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8207 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8208 mConfig.inputCfg.bufferProvider.cookie = NULL; 8209 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8210 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8211 mConfig.outputCfg.bufferProvider.cookie = NULL; 8212 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8213 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8214 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8215 // Insert effect: 8216 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8217 // always overwrites output buffer: input buffer == output buffer 8218 // - in other sessions: 8219 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8220 // other effect: overwrites output buffer: input buffer == output buffer 8221 // Auxiliary effect: 8222 // accumulates in output buffer: input buffer != output buffer 8223 // Therefore: accumulate <=> input buffer != output buffer 8224 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8225 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8226 } else { 8227 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8228 } 8229 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8230 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8231 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8232 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8233 8234 ALOGV("configure() %p thread %p buffer %p framecount %d", 8235 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8236 8237 status_t cmdStatus; 8238 uint32_t size = sizeof(int); 8239 status_t status = (*mEffectInterface)->command(mEffectInterface, 8240 EFFECT_CMD_SET_CONFIG, 8241 sizeof(effect_config_t), 8242 &mConfig, 8243 &size, 8244 &cmdStatus); 8245 if (status == 0) { 8246 status = cmdStatus; 8247 } 8248 8249 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8250 (1000 * mConfig.outputCfg.buffer.frameCount); 8251 8252 return status; 8253} 8254 8255status_t AudioFlinger::EffectModule::init() 8256{ 8257 Mutex::Autolock _l(mLock); 8258 if (mEffectInterface == NULL) { 8259 return NO_INIT; 8260 } 8261 status_t cmdStatus; 8262 uint32_t size = sizeof(status_t); 8263 status_t status = (*mEffectInterface)->command(mEffectInterface, 8264 EFFECT_CMD_INIT, 8265 0, 8266 NULL, 8267 &size, 8268 &cmdStatus); 8269 if (status == 0) { 8270 status = cmdStatus; 8271 } 8272 return status; 8273} 8274 8275status_t AudioFlinger::EffectModule::start() 8276{ 8277 Mutex::Autolock _l(mLock); 8278 return start_l(); 8279} 8280 8281status_t AudioFlinger::EffectModule::start_l() 8282{ 8283 if (mEffectInterface == NULL) { 8284 return NO_INIT; 8285 } 8286 status_t cmdStatus; 8287 uint32_t size = sizeof(status_t); 8288 status_t status = (*mEffectInterface)->command(mEffectInterface, 8289 EFFECT_CMD_ENABLE, 8290 0, 8291 NULL, 8292 &size, 8293 &cmdStatus); 8294 if (status == 0) { 8295 status = cmdStatus; 8296 } 8297 if (status == 0 && 8298 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8299 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8300 sp<ThreadBase> thread = mThread.promote(); 8301 if (thread != 0) { 8302 audio_stream_t *stream = thread->stream(); 8303 if (stream != NULL) { 8304 stream->add_audio_effect(stream, mEffectInterface); 8305 } 8306 } 8307 } 8308 return status; 8309} 8310 8311status_t AudioFlinger::EffectModule::stop() 8312{ 8313 Mutex::Autolock _l(mLock); 8314 return stop_l(); 8315} 8316 8317status_t AudioFlinger::EffectModule::stop_l() 8318{ 8319 if (mEffectInterface == NULL) { 8320 return NO_INIT; 8321 } 8322 status_t cmdStatus; 8323 uint32_t size = sizeof(status_t); 8324 status_t status = (*mEffectInterface)->command(mEffectInterface, 8325 EFFECT_CMD_DISABLE, 8326 0, 8327 NULL, 8328 &size, 8329 &cmdStatus); 8330 if (status == 0) { 8331 status = cmdStatus; 8332 } 8333 if (status == 0 && 8334 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8335 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8336 sp<ThreadBase> thread = mThread.promote(); 8337 if (thread != 0) { 8338 audio_stream_t *stream = thread->stream(); 8339 if (stream != NULL) { 8340 stream->remove_audio_effect(stream, mEffectInterface); 8341 } 8342 } 8343 } 8344 return status; 8345} 8346 8347status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8348 uint32_t cmdSize, 8349 void *pCmdData, 8350 uint32_t *replySize, 8351 void *pReplyData) 8352{ 8353 Mutex::Autolock _l(mLock); 8354// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8355 8356 if (mState == DESTROYED || mEffectInterface == NULL) { 8357 return NO_INIT; 8358 } 8359 status_t status = (*mEffectInterface)->command(mEffectInterface, 8360 cmdCode, 8361 cmdSize, 8362 pCmdData, 8363 replySize, 8364 pReplyData); 8365 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8366 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8367 for (size_t i = 1; i < mHandles.size(); i++) { 8368 sp<EffectHandle> h = mHandles[i].promote(); 8369 if (h != 0) { 8370 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8371 } 8372 } 8373 } 8374 return status; 8375} 8376 8377status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8378{ 8379 8380 Mutex::Autolock _l(mLock); 8381 ALOGV("setEnabled %p enabled %d", this, enabled); 8382 8383 if (enabled != isEnabled()) { 8384 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8385 if (enabled && status != NO_ERROR) { 8386 return status; 8387 } 8388 8389 switch (mState) { 8390 // going from disabled to enabled 8391 case IDLE: 8392 mState = STARTING; 8393 break; 8394 case STOPPED: 8395 mState = RESTART; 8396 break; 8397 case STOPPING: 8398 mState = ACTIVE; 8399 break; 8400 8401 // going from enabled to disabled 8402 case RESTART: 8403 mState = STOPPED; 8404 break; 8405 case STARTING: 8406 mState = IDLE; 8407 break; 8408 case ACTIVE: 8409 mState = STOPPING; 8410 break; 8411 case DESTROYED: 8412 return NO_ERROR; // simply ignore as we are being destroyed 8413 } 8414 for (size_t i = 1; i < mHandles.size(); i++) { 8415 sp<EffectHandle> h = mHandles[i].promote(); 8416 if (h != 0) { 8417 h->setEnabled(enabled); 8418 } 8419 } 8420 } 8421 return NO_ERROR; 8422} 8423 8424bool AudioFlinger::EffectModule::isEnabled() const 8425{ 8426 switch (mState) { 8427 case RESTART: 8428 case STARTING: 8429 case ACTIVE: 8430 return true; 8431 case IDLE: 8432 case STOPPING: 8433 case STOPPED: 8434 case DESTROYED: 8435 default: 8436 return false; 8437 } 8438} 8439 8440bool AudioFlinger::EffectModule::isProcessEnabled() const 8441{ 8442 switch (mState) { 8443 case RESTART: 8444 case ACTIVE: 8445 case STOPPING: 8446 case STOPPED: 8447 return true; 8448 case IDLE: 8449 case STARTING: 8450 case DESTROYED: 8451 default: 8452 return false; 8453 } 8454} 8455 8456status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8457{ 8458 Mutex::Autolock _l(mLock); 8459 status_t status = NO_ERROR; 8460 8461 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8462 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8463 if (isProcessEnabled() && 8464 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8465 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8466 status_t cmdStatus; 8467 uint32_t volume[2]; 8468 uint32_t *pVolume = NULL; 8469 uint32_t size = sizeof(volume); 8470 volume[0] = *left; 8471 volume[1] = *right; 8472 if (controller) { 8473 pVolume = volume; 8474 } 8475 status = (*mEffectInterface)->command(mEffectInterface, 8476 EFFECT_CMD_SET_VOLUME, 8477 size, 8478 volume, 8479 &size, 8480 pVolume); 8481 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8482 *left = volume[0]; 8483 *right = volume[1]; 8484 } 8485 } 8486 return status; 8487} 8488 8489status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8490{ 8491 Mutex::Autolock _l(mLock); 8492 status_t status = NO_ERROR; 8493 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8494 // audio pre processing modules on RecordThread can receive both output and 8495 // input device indication in the same call 8496 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8497 if (dev) { 8498 status_t cmdStatus; 8499 uint32_t size = sizeof(status_t); 8500 8501 status = (*mEffectInterface)->command(mEffectInterface, 8502 EFFECT_CMD_SET_DEVICE, 8503 sizeof(uint32_t), 8504 &dev, 8505 &size, 8506 &cmdStatus); 8507 if (status == NO_ERROR) { 8508 status = cmdStatus; 8509 } 8510 } 8511 dev = device & AUDIO_DEVICE_IN_ALL; 8512 if (dev) { 8513 status_t cmdStatus; 8514 uint32_t size = sizeof(status_t); 8515 8516 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8517 EFFECT_CMD_SET_INPUT_DEVICE, 8518 sizeof(uint32_t), 8519 &dev, 8520 &size, 8521 &cmdStatus); 8522 if (status2 == NO_ERROR) { 8523 status2 = cmdStatus; 8524 } 8525 if (status == NO_ERROR) { 8526 status = status2; 8527 } 8528 } 8529 } 8530 return status; 8531} 8532 8533status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8534{ 8535 Mutex::Autolock _l(mLock); 8536 status_t status = NO_ERROR; 8537 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8538 status_t cmdStatus; 8539 uint32_t size = sizeof(status_t); 8540 status = (*mEffectInterface)->command(mEffectInterface, 8541 EFFECT_CMD_SET_AUDIO_MODE, 8542 sizeof(audio_mode_t), 8543 &mode, 8544 &size, 8545 &cmdStatus); 8546 if (status == NO_ERROR) { 8547 status = cmdStatus; 8548 } 8549 } 8550 return status; 8551} 8552 8553void AudioFlinger::EffectModule::setSuspended(bool suspended) 8554{ 8555 Mutex::Autolock _l(mLock); 8556 mSuspended = suspended; 8557} 8558 8559bool AudioFlinger::EffectModule::suspended() const 8560{ 8561 Mutex::Autolock _l(mLock); 8562 return mSuspended; 8563} 8564 8565status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8566{ 8567 const size_t SIZE = 256; 8568 char buffer[SIZE]; 8569 String8 result; 8570 8571 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8572 result.append(buffer); 8573 8574 bool locked = tryLock(mLock); 8575 // failed to lock - AudioFlinger is probably deadlocked 8576 if (!locked) { 8577 result.append("\t\tCould not lock Fx mutex:\n"); 8578 } 8579 8580 result.append("\t\tSession Status State Engine:\n"); 8581 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8582 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8583 result.append(buffer); 8584 8585 result.append("\t\tDescriptor:\n"); 8586 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8587 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8588 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8589 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8590 result.append(buffer); 8591 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8592 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8593 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8594 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8595 result.append(buffer); 8596 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8597 mDescriptor.apiVersion, 8598 mDescriptor.flags); 8599 result.append(buffer); 8600 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8601 mDescriptor.name); 8602 result.append(buffer); 8603 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8604 mDescriptor.implementor); 8605 result.append(buffer); 8606 8607 result.append("\t\t- Input configuration:\n"); 8608 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8609 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8610 (uint32_t)mConfig.inputCfg.buffer.raw, 8611 mConfig.inputCfg.buffer.frameCount, 8612 mConfig.inputCfg.samplingRate, 8613 mConfig.inputCfg.channels, 8614 mConfig.inputCfg.format); 8615 result.append(buffer); 8616 8617 result.append("\t\t- Output configuration:\n"); 8618 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8619 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8620 (uint32_t)mConfig.outputCfg.buffer.raw, 8621 mConfig.outputCfg.buffer.frameCount, 8622 mConfig.outputCfg.samplingRate, 8623 mConfig.outputCfg.channels, 8624 mConfig.outputCfg.format); 8625 result.append(buffer); 8626 8627 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8628 result.append(buffer); 8629 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8630 for (size_t i = 0; i < mHandles.size(); ++i) { 8631 sp<EffectHandle> handle = mHandles[i].promote(); 8632 if (handle != 0) { 8633 handle->dump(buffer, SIZE); 8634 result.append(buffer); 8635 } 8636 } 8637 8638 result.append("\n"); 8639 8640 write(fd, result.string(), result.length()); 8641 8642 if (locked) { 8643 mLock.unlock(); 8644 } 8645 8646 return NO_ERROR; 8647} 8648 8649// ---------------------------------------------------------------------------- 8650// EffectHandle implementation 8651// ---------------------------------------------------------------------------- 8652 8653#undef LOG_TAG 8654#define LOG_TAG "AudioFlinger::EffectHandle" 8655 8656AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8657 const sp<AudioFlinger::Client>& client, 8658 const sp<IEffectClient>& effectClient, 8659 int32_t priority) 8660 : BnEffect(), 8661 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8662 mPriority(priority), mHasControl(false), mEnabled(false) 8663{ 8664 ALOGV("constructor %p", this); 8665 8666 if (client == 0) { 8667 return; 8668 } 8669 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8670 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8671 if (mCblkMemory != 0) { 8672 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8673 8674 if (mCblk != NULL) { 8675 new(mCblk) effect_param_cblk_t(); 8676 mBuffer = (uint8_t *)mCblk + bufOffset; 8677 } 8678 } else { 8679 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8680 return; 8681 } 8682} 8683 8684AudioFlinger::EffectHandle::~EffectHandle() 8685{ 8686 ALOGV("Destructor %p", this); 8687 disconnect(false); 8688 ALOGV("Destructor DONE %p", this); 8689} 8690 8691status_t AudioFlinger::EffectHandle::enable() 8692{ 8693 ALOGV("enable %p", this); 8694 if (!mHasControl) return INVALID_OPERATION; 8695 if (mEffect == 0) return DEAD_OBJECT; 8696 8697 if (mEnabled) { 8698 return NO_ERROR; 8699 } 8700 8701 mEnabled = true; 8702 8703 sp<ThreadBase> thread = mEffect->thread().promote(); 8704 if (thread != 0) { 8705 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8706 } 8707 8708 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8709 if (mEffect->suspended()) { 8710 return NO_ERROR; 8711 } 8712 8713 status_t status = mEffect->setEnabled(true); 8714 if (status != NO_ERROR) { 8715 if (thread != 0) { 8716 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8717 } 8718 mEnabled = false; 8719 } 8720 return status; 8721} 8722 8723status_t AudioFlinger::EffectHandle::disable() 8724{ 8725 ALOGV("disable %p", this); 8726 if (!mHasControl) return INVALID_OPERATION; 8727 if (mEffect == 0) return DEAD_OBJECT; 8728 8729 if (!mEnabled) { 8730 return NO_ERROR; 8731 } 8732 mEnabled = false; 8733 8734 if (mEffect->suspended()) { 8735 return NO_ERROR; 8736 } 8737 8738 status_t status = mEffect->setEnabled(false); 8739 8740 sp<ThreadBase> thread = mEffect->thread().promote(); 8741 if (thread != 0) { 8742 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8743 } 8744 8745 return status; 8746} 8747 8748void AudioFlinger::EffectHandle::disconnect() 8749{ 8750 disconnect(true); 8751} 8752 8753void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8754{ 8755 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8756 if (mEffect == 0) { 8757 return; 8758 } 8759 mEffect->disconnect(this, unpinIfLast); 8760 8761 if (mHasControl && mEnabled) { 8762 sp<ThreadBase> thread = mEffect->thread().promote(); 8763 if (thread != 0) { 8764 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8765 } 8766 } 8767 8768 // release sp on module => module destructor can be called now 8769 mEffect.clear(); 8770 if (mClient != 0) { 8771 if (mCblk != NULL) { 8772 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8773 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8774 } 8775 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8776 // Client destructor must run with AudioFlinger mutex locked 8777 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8778 mClient.clear(); 8779 } 8780} 8781 8782status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8783 uint32_t cmdSize, 8784 void *pCmdData, 8785 uint32_t *replySize, 8786 void *pReplyData) 8787{ 8788// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8789// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8790 8791 // only get parameter command is permitted for applications not controlling the effect 8792 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8793 return INVALID_OPERATION; 8794 } 8795 if (mEffect == 0) return DEAD_OBJECT; 8796 if (mClient == 0) return INVALID_OPERATION; 8797 8798 // handle commands that are not forwarded transparently to effect engine 8799 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8800 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8801 // no risk to block the whole media server process or mixer threads is we are stuck here 8802 Mutex::Autolock _l(mCblk->lock); 8803 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8804 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8805 mCblk->serverIndex = 0; 8806 mCblk->clientIndex = 0; 8807 return BAD_VALUE; 8808 } 8809 status_t status = NO_ERROR; 8810 while (mCblk->serverIndex < mCblk->clientIndex) { 8811 int reply; 8812 uint32_t rsize = sizeof(int); 8813 int *p = (int *)(mBuffer + mCblk->serverIndex); 8814 int size = *p++; 8815 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8816 ALOGW("command(): invalid parameter block size"); 8817 break; 8818 } 8819 effect_param_t *param = (effect_param_t *)p; 8820 if (param->psize == 0 || param->vsize == 0) { 8821 ALOGW("command(): null parameter or value size"); 8822 mCblk->serverIndex += size; 8823 continue; 8824 } 8825 uint32_t psize = sizeof(effect_param_t) + 8826 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8827 param->vsize; 8828 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8829 psize, 8830 p, 8831 &rsize, 8832 &reply); 8833 // stop at first error encountered 8834 if (ret != NO_ERROR) { 8835 status = ret; 8836 *(int *)pReplyData = reply; 8837 break; 8838 } else if (reply != NO_ERROR) { 8839 *(int *)pReplyData = reply; 8840 break; 8841 } 8842 mCblk->serverIndex += size; 8843 } 8844 mCblk->serverIndex = 0; 8845 mCblk->clientIndex = 0; 8846 return status; 8847 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8848 *(int *)pReplyData = NO_ERROR; 8849 return enable(); 8850 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8851 *(int *)pReplyData = NO_ERROR; 8852 return disable(); 8853 } 8854 8855 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8856} 8857 8858void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8859{ 8860 ALOGV("setControl %p control %d", this, hasControl); 8861 8862 mHasControl = hasControl; 8863 mEnabled = enabled; 8864 8865 if (signal && mEffectClient != 0) { 8866 mEffectClient->controlStatusChanged(hasControl); 8867 } 8868} 8869 8870void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8871 uint32_t cmdSize, 8872 void *pCmdData, 8873 uint32_t replySize, 8874 void *pReplyData) 8875{ 8876 if (mEffectClient != 0) { 8877 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8878 } 8879} 8880 8881 8882 8883void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8884{ 8885 if (mEffectClient != 0) { 8886 mEffectClient->enableStatusChanged(enabled); 8887 } 8888} 8889 8890status_t AudioFlinger::EffectHandle::onTransact( 8891 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8892{ 8893 return BnEffect::onTransact(code, data, reply, flags); 8894} 8895 8896 8897void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8898{ 8899 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8900 8901 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8902 (mClient == 0) ? getpid_cached : mClient->pid(), 8903 mPriority, 8904 mHasControl, 8905 !locked, 8906 mCblk ? mCblk->clientIndex : 0, 8907 mCblk ? mCblk->serverIndex : 0 8908 ); 8909 8910 if (locked) { 8911 mCblk->lock.unlock(); 8912 } 8913} 8914 8915#undef LOG_TAG 8916#define LOG_TAG "AudioFlinger::EffectChain" 8917 8918AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8919 int sessionId) 8920 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8921 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8922 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8923{ 8924 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8925 if (thread == NULL) { 8926 return; 8927 } 8928 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8929 thread->frameCount(); 8930} 8931 8932AudioFlinger::EffectChain::~EffectChain() 8933{ 8934 if (mOwnInBuffer) { 8935 delete mInBuffer; 8936 } 8937 8938} 8939 8940// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8941sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8942{ 8943 size_t size = mEffects.size(); 8944 8945 for (size_t i = 0; i < size; i++) { 8946 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8947 return mEffects[i]; 8948 } 8949 } 8950 return 0; 8951} 8952 8953// getEffectFromId_l() must be called with ThreadBase::mLock held 8954sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8955{ 8956 size_t size = mEffects.size(); 8957 8958 for (size_t i = 0; i < size; i++) { 8959 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8960 if (id == 0 || mEffects[i]->id() == id) { 8961 return mEffects[i]; 8962 } 8963 } 8964 return 0; 8965} 8966 8967// getEffectFromType_l() must be called with ThreadBase::mLock held 8968sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8969 const effect_uuid_t *type) 8970{ 8971 size_t size = mEffects.size(); 8972 8973 for (size_t i = 0; i < size; i++) { 8974 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8975 return mEffects[i]; 8976 } 8977 } 8978 return 0; 8979} 8980 8981// Must be called with EffectChain::mLock locked 8982void AudioFlinger::EffectChain::process_l() 8983{ 8984 sp<ThreadBase> thread = mThread.promote(); 8985 if (thread == 0) { 8986 ALOGW("process_l(): cannot promote mixer thread"); 8987 return; 8988 } 8989 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8990 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8991 // always process effects unless no more tracks are on the session and the effect tail 8992 // has been rendered 8993 bool doProcess = true; 8994 if (!isGlobalSession) { 8995 bool tracksOnSession = (trackCnt() != 0); 8996 8997 if (!tracksOnSession && mTailBufferCount == 0) { 8998 doProcess = false; 8999 } 9000 9001 if (activeTrackCnt() == 0) { 9002 // if no track is active and the effect tail has not been rendered, 9003 // the input buffer must be cleared here as the mixer process will not do it 9004 if (tracksOnSession || mTailBufferCount > 0) { 9005 size_t numSamples = thread->frameCount() * thread->channelCount(); 9006 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9007 if (mTailBufferCount > 0) { 9008 mTailBufferCount--; 9009 } 9010 } 9011 } 9012 } 9013 9014 size_t size = mEffects.size(); 9015 if (doProcess) { 9016 for (size_t i = 0; i < size; i++) { 9017 mEffects[i]->process(); 9018 } 9019 } 9020 for (size_t i = 0; i < size; i++) { 9021 mEffects[i]->updateState(); 9022 } 9023} 9024 9025// addEffect_l() must be called with PlaybackThread::mLock held 9026status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9027{ 9028 effect_descriptor_t desc = effect->desc(); 9029 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9030 9031 Mutex::Autolock _l(mLock); 9032 effect->setChain(this); 9033 sp<ThreadBase> thread = mThread.promote(); 9034 if (thread == 0) { 9035 return NO_INIT; 9036 } 9037 effect->setThread(thread); 9038 9039 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9040 // Auxiliary effects are inserted at the beginning of mEffects vector as 9041 // they are processed first and accumulated in chain input buffer 9042 mEffects.insertAt(effect, 0); 9043 9044 // the input buffer for auxiliary effect contains mono samples in 9045 // 32 bit format. This is to avoid saturation in AudoMixer 9046 // accumulation stage. Saturation is done in EffectModule::process() before 9047 // calling the process in effect engine 9048 size_t numSamples = thread->frameCount(); 9049 int32_t *buffer = new int32_t[numSamples]; 9050 memset(buffer, 0, numSamples * sizeof(int32_t)); 9051 effect->setInBuffer((int16_t *)buffer); 9052 // auxiliary effects output samples to chain input buffer for further processing 9053 // by insert effects 9054 effect->setOutBuffer(mInBuffer); 9055 } else { 9056 // Insert effects are inserted at the end of mEffects vector as they are processed 9057 // after track and auxiliary effects. 9058 // Insert effect order as a function of indicated preference: 9059 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9060 // another effect is present 9061 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9062 // last effect claiming first position 9063 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9064 // first effect claiming last position 9065 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9066 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9067 // already present 9068 9069 size_t size = mEffects.size(); 9070 size_t idx_insert = size; 9071 ssize_t idx_insert_first = -1; 9072 ssize_t idx_insert_last = -1; 9073 9074 for (size_t i = 0; i < size; i++) { 9075 effect_descriptor_t d = mEffects[i]->desc(); 9076 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9077 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9078 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9079 // check invalid effect chaining combinations 9080 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9081 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9082 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9083 return INVALID_OPERATION; 9084 } 9085 // remember position of first insert effect and by default 9086 // select this as insert position for new effect 9087 if (idx_insert == size) { 9088 idx_insert = i; 9089 } 9090 // remember position of last insert effect claiming 9091 // first position 9092 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9093 idx_insert_first = i; 9094 } 9095 // remember position of first insert effect claiming 9096 // last position 9097 if (iPref == EFFECT_FLAG_INSERT_LAST && 9098 idx_insert_last == -1) { 9099 idx_insert_last = i; 9100 } 9101 } 9102 } 9103 9104 // modify idx_insert from first position if needed 9105 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9106 if (idx_insert_last != -1) { 9107 idx_insert = idx_insert_last; 9108 } else { 9109 idx_insert = size; 9110 } 9111 } else { 9112 if (idx_insert_first != -1) { 9113 idx_insert = idx_insert_first + 1; 9114 } 9115 } 9116 9117 // always read samples from chain input buffer 9118 effect->setInBuffer(mInBuffer); 9119 9120 // if last effect in the chain, output samples to chain 9121 // output buffer, otherwise to chain input buffer 9122 if (idx_insert == size) { 9123 if (idx_insert != 0) { 9124 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9125 mEffects[idx_insert-1]->configure(); 9126 } 9127 effect->setOutBuffer(mOutBuffer); 9128 } else { 9129 effect->setOutBuffer(mInBuffer); 9130 } 9131 mEffects.insertAt(effect, idx_insert); 9132 9133 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9134 } 9135 effect->configure(); 9136 return NO_ERROR; 9137} 9138 9139// removeEffect_l() must be called with PlaybackThread::mLock held 9140size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9141{ 9142 Mutex::Autolock _l(mLock); 9143 size_t size = mEffects.size(); 9144 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9145 9146 for (size_t i = 0; i < size; i++) { 9147 if (effect == mEffects[i]) { 9148 // calling stop here will remove pre-processing effect from the audio HAL. 9149 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9150 // the middle of a read from audio HAL 9151 if (mEffects[i]->state() == EffectModule::ACTIVE || 9152 mEffects[i]->state() == EffectModule::STOPPING) { 9153 mEffects[i]->stop(); 9154 } 9155 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9156 delete[] effect->inBuffer(); 9157 } else { 9158 if (i == size - 1 && i != 0) { 9159 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9160 mEffects[i - 1]->configure(); 9161 } 9162 } 9163 mEffects.removeAt(i); 9164 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9165 break; 9166 } 9167 } 9168 9169 return mEffects.size(); 9170} 9171 9172// setDevice_l() must be called with PlaybackThread::mLock held 9173void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9174{ 9175 size_t size = mEffects.size(); 9176 for (size_t i = 0; i < size; i++) { 9177 mEffects[i]->setDevice(device); 9178 } 9179} 9180 9181// setMode_l() must be called with PlaybackThread::mLock held 9182void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9183{ 9184 size_t size = mEffects.size(); 9185 for (size_t i = 0; i < size; i++) { 9186 mEffects[i]->setMode(mode); 9187 } 9188} 9189 9190// setVolume_l() must be called with PlaybackThread::mLock held 9191bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9192{ 9193 uint32_t newLeft = *left; 9194 uint32_t newRight = *right; 9195 bool hasControl = false; 9196 int ctrlIdx = -1; 9197 size_t size = mEffects.size(); 9198 9199 // first update volume controller 9200 for (size_t i = size; i > 0; i--) { 9201 if (mEffects[i - 1]->isProcessEnabled() && 9202 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9203 ctrlIdx = i - 1; 9204 hasControl = true; 9205 break; 9206 } 9207 } 9208 9209 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9210 if (hasControl) { 9211 *left = mNewLeftVolume; 9212 *right = mNewRightVolume; 9213 } 9214 return hasControl; 9215 } 9216 9217 mVolumeCtrlIdx = ctrlIdx; 9218 mLeftVolume = newLeft; 9219 mRightVolume = newRight; 9220 9221 // second get volume update from volume controller 9222 if (ctrlIdx >= 0) { 9223 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9224 mNewLeftVolume = newLeft; 9225 mNewRightVolume = newRight; 9226 } 9227 // then indicate volume to all other effects in chain. 9228 // Pass altered volume to effects before volume controller 9229 // and requested volume to effects after controller 9230 uint32_t lVol = newLeft; 9231 uint32_t rVol = newRight; 9232 9233 for (size_t i = 0; i < size; i++) { 9234 if ((int)i == ctrlIdx) continue; 9235 // this also works for ctrlIdx == -1 when there is no volume controller 9236 if ((int)i > ctrlIdx) { 9237 lVol = *left; 9238 rVol = *right; 9239 } 9240 mEffects[i]->setVolume(&lVol, &rVol, false); 9241 } 9242 *left = newLeft; 9243 *right = newRight; 9244 9245 return hasControl; 9246} 9247 9248status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9249{ 9250 const size_t SIZE = 256; 9251 char buffer[SIZE]; 9252 String8 result; 9253 9254 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9255 result.append(buffer); 9256 9257 bool locked = tryLock(mLock); 9258 // failed to lock - AudioFlinger is probably deadlocked 9259 if (!locked) { 9260 result.append("\tCould not lock mutex:\n"); 9261 } 9262 9263 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9264 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9265 mEffects.size(), 9266 (uint32_t)mInBuffer, 9267 (uint32_t)mOutBuffer, 9268 mActiveTrackCnt); 9269 result.append(buffer); 9270 write(fd, result.string(), result.size()); 9271 9272 for (size_t i = 0; i < mEffects.size(); ++i) { 9273 sp<EffectModule> effect = mEffects[i]; 9274 if (effect != 0) { 9275 effect->dump(fd, args); 9276 } 9277 } 9278 9279 if (locked) { 9280 mLock.unlock(); 9281 } 9282 9283 return NO_ERROR; 9284} 9285 9286// must be called with ThreadBase::mLock held 9287void AudioFlinger::EffectChain::setEffectSuspended_l( 9288 const effect_uuid_t *type, bool suspend) 9289{ 9290 sp<SuspendedEffectDesc> desc; 9291 // use effect type UUID timelow as key as there is no real risk of identical 9292 // timeLow fields among effect type UUIDs. 9293 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9294 if (suspend) { 9295 if (index >= 0) { 9296 desc = mSuspendedEffects.valueAt(index); 9297 } else { 9298 desc = new SuspendedEffectDesc(); 9299 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9300 mSuspendedEffects.add(type->timeLow, desc); 9301 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9302 } 9303 if (desc->mRefCount++ == 0) { 9304 sp<EffectModule> effect = getEffectIfEnabled(type); 9305 if (effect != 0) { 9306 desc->mEffect = effect; 9307 effect->setSuspended(true); 9308 effect->setEnabled(false); 9309 } 9310 } 9311 } else { 9312 if (index < 0) { 9313 return; 9314 } 9315 desc = mSuspendedEffects.valueAt(index); 9316 if (desc->mRefCount <= 0) { 9317 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9318 desc->mRefCount = 1; 9319 } 9320 if (--desc->mRefCount == 0) { 9321 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9322 if (desc->mEffect != 0) { 9323 sp<EffectModule> effect = desc->mEffect.promote(); 9324 if (effect != 0) { 9325 effect->setSuspended(false); 9326 sp<EffectHandle> handle = effect->controlHandle(); 9327 if (handle != 0) { 9328 effect->setEnabled(handle->enabled()); 9329 } 9330 } 9331 desc->mEffect.clear(); 9332 } 9333 mSuspendedEffects.removeItemsAt(index); 9334 } 9335 } 9336} 9337 9338// must be called with ThreadBase::mLock held 9339void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9340{ 9341 sp<SuspendedEffectDesc> desc; 9342 9343 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9344 if (suspend) { 9345 if (index >= 0) { 9346 desc = mSuspendedEffects.valueAt(index); 9347 } else { 9348 desc = new SuspendedEffectDesc(); 9349 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9350 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9351 } 9352 if (desc->mRefCount++ == 0) { 9353 Vector< sp<EffectModule> > effects; 9354 getSuspendEligibleEffects(effects); 9355 for (size_t i = 0; i < effects.size(); i++) { 9356 setEffectSuspended_l(&effects[i]->desc().type, true); 9357 } 9358 } 9359 } else { 9360 if (index < 0) { 9361 return; 9362 } 9363 desc = mSuspendedEffects.valueAt(index); 9364 if (desc->mRefCount <= 0) { 9365 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9366 desc->mRefCount = 1; 9367 } 9368 if (--desc->mRefCount == 0) { 9369 Vector<const effect_uuid_t *> types; 9370 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9371 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9372 continue; 9373 } 9374 types.add(&mSuspendedEffects.valueAt(i)->mType); 9375 } 9376 for (size_t i = 0; i < types.size(); i++) { 9377 setEffectSuspended_l(types[i], false); 9378 } 9379 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9380 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9381 } 9382 } 9383} 9384 9385 9386// The volume effect is used for automated tests only 9387#ifndef OPENSL_ES_H_ 9388static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9389 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9390const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9391#endif //OPENSL_ES_H_ 9392 9393bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9394{ 9395 // auxiliary effects and visualizer are never suspended on output mix 9396 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9397 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9398 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9399 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9400 return false; 9401 } 9402 return true; 9403} 9404 9405void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9406{ 9407 effects.clear(); 9408 for (size_t i = 0; i < mEffects.size(); i++) { 9409 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9410 effects.add(mEffects[i]); 9411 } 9412 } 9413} 9414 9415sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9416 const effect_uuid_t *type) 9417{ 9418 sp<EffectModule> effect = getEffectFromType_l(type); 9419 return effect != 0 && effect->isEnabled() ? effect : 0; 9420} 9421 9422void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9423 bool enabled) 9424{ 9425 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9426 if (enabled) { 9427 if (index < 0) { 9428 // if the effect is not suspend check if all effects are suspended 9429 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9430 if (index < 0) { 9431 return; 9432 } 9433 if (!isEffectEligibleForSuspend(effect->desc())) { 9434 return; 9435 } 9436 setEffectSuspended_l(&effect->desc().type, enabled); 9437 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9438 if (index < 0) { 9439 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9440 return; 9441 } 9442 } 9443 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9444 effect->desc().type.timeLow); 9445 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9446 // if effect is requested to suspended but was not yet enabled, supend it now. 9447 if (desc->mEffect == 0) { 9448 desc->mEffect = effect; 9449 effect->setEnabled(false); 9450 effect->setSuspended(true); 9451 } 9452 } else { 9453 if (index < 0) { 9454 return; 9455 } 9456 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9457 effect->desc().type.timeLow); 9458 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9459 desc->mEffect.clear(); 9460 effect->setSuspended(false); 9461 } 9462} 9463 9464#undef LOG_TAG 9465#define LOG_TAG "AudioFlinger" 9466 9467// ---------------------------------------------------------------------------- 9468 9469status_t AudioFlinger::onTransact( 9470 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9471{ 9472 return BnAudioFlinger::onTransact(code, data, reply, flags); 9473} 9474 9475}; // namespace android 9476