AudioFlinger.cpp revision 3acbd053c842e76e1a40fc8a0bf62de87eebf00f
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#undef ADD_BATTERY_DATA 41 42#ifdef ADD_BATTERY_DATA 43#include <media/IMediaPlayerService.h> 44#include <media/IMediaDeathNotifier.h> 45#endif 46 47#include <private/media/AudioTrackShared.h> 48#include <private/media/AudioEffectShared.h> 49 50#include <system/audio.h> 51#include <hardware/audio.h> 52 53#include "AudioMixer.h" 54#include "AudioFlinger.h" 55#include "ServiceUtilities.h" 56 57#include <media/EffectsFactoryApi.h> 58#include <audio_effects/effect_visualizer.h> 59#include <audio_effects/effect_ns.h> 60#include <audio_effects/effect_aec.h> 61 62#include <audio_utils/primitives.h> 63 64#include <powermanager/PowerManager.h> 65 66// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72#include <common_time/cc_helper.h> 73#include <common_time/local_clock.h> 74 75// ---------------------------------------------------------------------------- 76 77// Note: the following macro is used for extremely verbose logging message. In 78// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 79// 0; but one side effect of this is to turn all LOGV's as well. Some messages 80// are so verbose that we want to suppress them even when we have ALOG_ASSERT 81// turned on. Do not uncomment the #def below unless you really know what you 82// are doing and want to see all of the extremely verbose messages. 83//#define VERY_VERY_VERBOSE_LOGGING 84#ifdef VERY_VERY_VERBOSE_LOGGING 85#define ALOGVV ALOGV 86#else 87#define ALOGVV(a...) do { } while(0) 88#endif 89 90namespace android { 91 92static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 93static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 94 95static const float MAX_GAIN = 4096.0f; 96static const uint32_t MAX_GAIN_INT = 0x1000; 97 98// retry counts for buffer fill timeout 99// 50 * ~20msecs = 1 second 100static const int8_t kMaxTrackRetries = 50; 101static const int8_t kMaxTrackStartupRetries = 50; 102// allow less retry attempts on direct output thread. 103// direct outputs can be a scarce resource in audio hardware and should 104// be released as quickly as possible. 105static const int8_t kMaxTrackRetriesDirect = 2; 106 107static const int kDumpLockRetries = 50; 108static const int kDumpLockSleepUs = 20000; 109 110// don't warn about blocked writes or record buffer overflows more often than this 111static const nsecs_t kWarningThrottleNs = seconds(5); 112 113// RecordThread loop sleep time upon application overrun or audio HAL read error 114static const int kRecordThreadSleepUs = 5000; 115 116// maximum time to wait for setParameters to complete 117static const nsecs_t kSetParametersTimeoutNs = seconds(2); 118 119// minimum sleep time for the mixer thread loop when tracks are active but in underrun 120static const uint32_t kMinThreadSleepTimeUs = 5000; 121// maximum divider applied to the active sleep time in the mixer thread loop 122static const uint32_t kMaxThreadSleepTimeShift = 2; 123 124nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 125 126// ---------------------------------------------------------------------------- 127 128#ifdef ADD_BATTERY_DATA 129// To collect the amplifier usage 130static void addBatteryData(uint32_t params) { 131 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 132 if (service == NULL) { 133 // it already logged 134 return; 135 } 136 137 service->addBatteryData(params); 138} 139#endif 140 141static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 142{ 143 const hw_module_t *mod; 144 int rc; 145 146 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 147 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 148 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 149 if (rc) { 150 goto out; 151 } 152 rc = audio_hw_device_open(mod, dev); 153 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 154 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 155 if (rc) { 156 goto out; 157 } 158 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 159 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 160 rc = BAD_VALUE; 161 goto out; 162 } 163 return 0; 164 165out: 166 *dev = NULL; 167 return rc; 168} 169 170// ---------------------------------------------------------------------------- 171 172AudioFlinger::AudioFlinger() 173 : BnAudioFlinger(), 174 mPrimaryHardwareDev(NULL), 175 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 176 mMasterVolume(1.0f), 177 mMasterVolumeSupportLvl(MVS_NONE), 178 mMasterMute(false), 179 mNextUniqueId(1), 180 mMode(AUDIO_MODE_INVALID), 181 mBtNrecIsOff(false) 182{ 183} 184 185void AudioFlinger::onFirstRef() 186{ 187 int rc = 0; 188 189 Mutex::Autolock _l(mLock); 190 191 /* TODO: move all this work into an Init() function */ 192 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 193 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 194 uint32_t int_val; 195 if (1 == sscanf(val_str, "%u", &int_val)) { 196 mStandbyTimeInNsecs = milliseconds(int_val); 197 ALOGI("Using %u mSec as standby time.", int_val); 198 } else { 199 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 200 ALOGI("Using default %u mSec as standby time.", 201 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 202 } 203 } 204 205 mMode = AUDIO_MODE_NORMAL; 206 mMasterVolumeSW = 1.0; 207 mMasterVolume = 1.0; 208 mHardwareStatus = AUDIO_HW_IDLE; 209} 210 211AudioFlinger::~AudioFlinger() 212{ 213 214 while (!mRecordThreads.isEmpty()) { 215 // closeInput() will remove first entry from mRecordThreads 216 closeInput(mRecordThreads.keyAt(0)); 217 } 218 while (!mPlaybackThreads.isEmpty()) { 219 // closeOutput() will remove first entry from mPlaybackThreads 220 closeOutput(mPlaybackThreads.keyAt(0)); 221 } 222 223 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 224 // no mHardwareLock needed, as there are no other references to this 225 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 226 delete mAudioHwDevs.valueAt(i); 227 } 228} 229 230static const char * const audio_interfaces[] = { 231 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 232 AUDIO_HARDWARE_MODULE_ID_A2DP, 233 AUDIO_HARDWARE_MODULE_ID_USB, 234}; 235#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 236 237audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 238{ 239 // if module is 0, the request comes from an old policy manager and we should load 240 // well known modules 241 if (module == 0) { 242 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 243 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 244 loadHwModule_l(audio_interfaces[i]); 245 } 246 } else { 247 // check a match for the requested module handle 248 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 249 if (audioHwdevice != NULL) { 250 return audioHwdevice->hwDevice(); 251 } 252 } 253 // then try to find a module supporting the requested device. 254 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 255 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 256 if ((dev->get_supported_devices(dev) & devices) == devices) 257 return dev; 258 } 259 260 return NULL; 261} 262 263status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 264{ 265 const size_t SIZE = 256; 266 char buffer[SIZE]; 267 String8 result; 268 269 result.append("Clients:\n"); 270 for (size_t i = 0; i < mClients.size(); ++i) { 271 sp<Client> client = mClients.valueAt(i).promote(); 272 if (client != 0) { 273 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 274 result.append(buffer); 275 } 276 } 277 278 result.append("Global session refs:\n"); 279 result.append(" session pid count\n"); 280 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 281 AudioSessionRef *r = mAudioSessionRefs[i]; 282 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 283 result.append(buffer); 284 } 285 write(fd, result.string(), result.size()); 286 return NO_ERROR; 287} 288 289 290status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 291{ 292 const size_t SIZE = 256; 293 char buffer[SIZE]; 294 String8 result; 295 hardware_call_state hardwareStatus = mHardwareStatus; 296 297 snprintf(buffer, SIZE, "Hardware status: %d\n" 298 "Standby Time mSec: %u\n", 299 hardwareStatus, 300 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 301 result.append(buffer); 302 write(fd, result.string(), result.size()); 303 return NO_ERROR; 304} 305 306status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 307{ 308 const size_t SIZE = 256; 309 char buffer[SIZE]; 310 String8 result; 311 snprintf(buffer, SIZE, "Permission Denial: " 312 "can't dump AudioFlinger from pid=%d, uid=%d\n", 313 IPCThreadState::self()->getCallingPid(), 314 IPCThreadState::self()->getCallingUid()); 315 result.append(buffer); 316 write(fd, result.string(), result.size()); 317 return NO_ERROR; 318} 319 320static bool tryLock(Mutex& mutex) 321{ 322 bool locked = false; 323 for (int i = 0; i < kDumpLockRetries; ++i) { 324 if (mutex.tryLock() == NO_ERROR) { 325 locked = true; 326 break; 327 } 328 usleep(kDumpLockSleepUs); 329 } 330 return locked; 331} 332 333status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 334{ 335 if (!dumpAllowed()) { 336 dumpPermissionDenial(fd, args); 337 } else { 338 // get state of hardware lock 339 bool hardwareLocked = tryLock(mHardwareLock); 340 if (!hardwareLocked) { 341 String8 result(kHardwareLockedString); 342 write(fd, result.string(), result.size()); 343 } else { 344 mHardwareLock.unlock(); 345 } 346 347 bool locked = tryLock(mLock); 348 349 // failed to lock - AudioFlinger is probably deadlocked 350 if (!locked) { 351 String8 result(kDeadlockedString); 352 write(fd, result.string(), result.size()); 353 } 354 355 dumpClients(fd, args); 356 dumpInternals(fd, args); 357 358 // dump playback threads 359 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 360 mPlaybackThreads.valueAt(i)->dump(fd, args); 361 } 362 363 // dump record threads 364 for (size_t i = 0; i < mRecordThreads.size(); i++) { 365 mRecordThreads.valueAt(i)->dump(fd, args); 366 } 367 368 // dump all hardware devs 369 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 370 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 371 dev->dump(dev, fd); 372 } 373 if (locked) mLock.unlock(); 374 } 375 return NO_ERROR; 376} 377 378sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 379{ 380 // If pid is already in the mClients wp<> map, then use that entry 381 // (for which promote() is always != 0), otherwise create a new entry and Client. 382 sp<Client> client = mClients.valueFor(pid).promote(); 383 if (client == 0) { 384 client = new Client(this, pid); 385 mClients.add(pid, client); 386 } 387 388 return client; 389} 390 391// IAudioFlinger interface 392 393 394sp<IAudioTrack> AudioFlinger::createTrack( 395 pid_t pid, 396 audio_stream_type_t streamType, 397 uint32_t sampleRate, 398 audio_format_t format, 399 uint32_t channelMask, 400 int frameCount, 401 IAudioFlinger::track_flags_t flags, 402 const sp<IMemory>& sharedBuffer, 403 audio_io_handle_t output, 404 pid_t tid, 405 int *sessionId, 406 status_t *status) 407{ 408 sp<PlaybackThread::Track> track; 409 sp<TrackHandle> trackHandle; 410 sp<Client> client; 411 status_t lStatus; 412 int lSessionId; 413 414 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 415 // but if someone uses binder directly they could bypass that and cause us to crash 416 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 417 ALOGE("createTrack() invalid stream type %d", streamType); 418 lStatus = BAD_VALUE; 419 goto Exit; 420 } 421 422 { 423 Mutex::Autolock _l(mLock); 424 PlaybackThread *thread = checkPlaybackThread_l(output); 425 PlaybackThread *effectThread = NULL; 426 if (thread == NULL) { 427 ALOGE("unknown output thread"); 428 lStatus = BAD_VALUE; 429 goto Exit; 430 } 431 432 client = registerPid_l(pid); 433 434 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 435 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 436 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 437 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 438 if (mPlaybackThreads.keyAt(i) != output) { 439 // prevent same audio session on different output threads 440 uint32_t sessions = t->hasAudioSession(*sessionId); 441 if (sessions & PlaybackThread::TRACK_SESSION) { 442 ALOGE("createTrack() session ID %d already in use", *sessionId); 443 lStatus = BAD_VALUE; 444 goto Exit; 445 } 446 // check if an effect with same session ID is waiting for a track to be created 447 if (sessions & PlaybackThread::EFFECT_SESSION) { 448 effectThread = t.get(); 449 } 450 } 451 } 452 lSessionId = *sessionId; 453 } else { 454 // if no audio session id is provided, create one here 455 lSessionId = nextUniqueId(); 456 if (sessionId != NULL) { 457 *sessionId = lSessionId; 458 } 459 } 460 ALOGV("createTrack() lSessionId: %d", lSessionId); 461 462 track = thread->createTrack_l(client, streamType, sampleRate, format, 463 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 464 465 // move effect chain to this output thread if an effect on same session was waiting 466 // for a track to be created 467 if (lStatus == NO_ERROR && effectThread != NULL) { 468 Mutex::Autolock _dl(thread->mLock); 469 Mutex::Autolock _sl(effectThread->mLock); 470 moveEffectChain_l(lSessionId, effectThread, thread, true); 471 } 472 473 // Look for sync events awaiting for a session to be used. 474 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 475 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 476 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 477 track->setSyncEvent(mPendingSyncEvents[i]); 478 mPendingSyncEvents.removeAt(i); 479 i--; 480 } 481 } 482 } 483 } 484 if (lStatus == NO_ERROR) { 485 trackHandle = new TrackHandle(track); 486 } else { 487 // remove local strong reference to Client before deleting the Track so that the Client 488 // destructor is called by the TrackBase destructor with mLock held 489 client.clear(); 490 track.clear(); 491 } 492 493Exit: 494 if (status != NULL) { 495 *status = lStatus; 496 } 497 return trackHandle; 498} 499 500uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 501{ 502 Mutex::Autolock _l(mLock); 503 PlaybackThread *thread = checkPlaybackThread_l(output); 504 if (thread == NULL) { 505 ALOGW("sampleRate() unknown thread %d", output); 506 return 0; 507 } 508 return thread->sampleRate(); 509} 510 511int AudioFlinger::channelCount(audio_io_handle_t output) const 512{ 513 Mutex::Autolock _l(mLock); 514 PlaybackThread *thread = checkPlaybackThread_l(output); 515 if (thread == NULL) { 516 ALOGW("channelCount() unknown thread %d", output); 517 return 0; 518 } 519 return thread->channelCount(); 520} 521 522audio_format_t AudioFlinger::format(audio_io_handle_t output) const 523{ 524 Mutex::Autolock _l(mLock); 525 PlaybackThread *thread = checkPlaybackThread_l(output); 526 if (thread == NULL) { 527 ALOGW("format() unknown thread %d", output); 528 return AUDIO_FORMAT_INVALID; 529 } 530 return thread->format(); 531} 532 533size_t AudioFlinger::frameCount(audio_io_handle_t output) const 534{ 535 Mutex::Autolock _l(mLock); 536 PlaybackThread *thread = checkPlaybackThread_l(output); 537 if (thread == NULL) { 538 ALOGW("frameCount() unknown thread %d", output); 539 return 0; 540 } 541 return thread->frameCount(); 542} 543 544uint32_t AudioFlinger::latency(audio_io_handle_t output) const 545{ 546 Mutex::Autolock _l(mLock); 547 PlaybackThread *thread = checkPlaybackThread_l(output); 548 if (thread == NULL) { 549 ALOGW("latency() unknown thread %d", output); 550 return 0; 551 } 552 return thread->latency(); 553} 554 555status_t AudioFlinger::setMasterVolume(float value) 556{ 557 status_t ret = initCheck(); 558 if (ret != NO_ERROR) { 559 return ret; 560 } 561 562 // check calling permissions 563 if (!settingsAllowed()) { 564 return PERMISSION_DENIED; 565 } 566 567 float swmv = value; 568 569 Mutex::Autolock _l(mLock); 570 571 // when hw supports master volume, don't scale in sw mixer 572 if (MVS_NONE != mMasterVolumeSupportLvl) { 573 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 574 AutoMutex lock(mHardwareLock); 575 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 576 577 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 578 if (NULL != dev->set_master_volume) { 579 dev->set_master_volume(dev, value); 580 } 581 mHardwareStatus = AUDIO_HW_IDLE; 582 } 583 584 swmv = 1.0; 585 } 586 587 mMasterVolume = value; 588 mMasterVolumeSW = swmv; 589 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 590 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 591 592 return NO_ERROR; 593} 594 595status_t AudioFlinger::setMode(audio_mode_t mode) 596{ 597 status_t ret = initCheck(); 598 if (ret != NO_ERROR) { 599 return ret; 600 } 601 602 // check calling permissions 603 if (!settingsAllowed()) { 604 return PERMISSION_DENIED; 605 } 606 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 607 ALOGW("Illegal value: setMode(%d)", mode); 608 return BAD_VALUE; 609 } 610 611 { // scope for the lock 612 AutoMutex lock(mHardwareLock); 613 mHardwareStatus = AUDIO_HW_SET_MODE; 614 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 615 mHardwareStatus = AUDIO_HW_IDLE; 616 } 617 618 if (NO_ERROR == ret) { 619 Mutex::Autolock _l(mLock); 620 mMode = mode; 621 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 622 mPlaybackThreads.valueAt(i)->setMode(mode); 623 } 624 625 return ret; 626} 627 628status_t AudioFlinger::setMicMute(bool state) 629{ 630 status_t ret = initCheck(); 631 if (ret != NO_ERROR) { 632 return ret; 633 } 634 635 // check calling permissions 636 if (!settingsAllowed()) { 637 return PERMISSION_DENIED; 638 } 639 640 AutoMutex lock(mHardwareLock); 641 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 642 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 643 mHardwareStatus = AUDIO_HW_IDLE; 644 return ret; 645} 646 647bool AudioFlinger::getMicMute() const 648{ 649 status_t ret = initCheck(); 650 if (ret != NO_ERROR) { 651 return false; 652 } 653 654 bool state = AUDIO_MODE_INVALID; 655 AutoMutex lock(mHardwareLock); 656 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 657 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 658 mHardwareStatus = AUDIO_HW_IDLE; 659 return state; 660} 661 662status_t AudioFlinger::setMasterMute(bool muted) 663{ 664 // check calling permissions 665 if (!settingsAllowed()) { 666 return PERMISSION_DENIED; 667 } 668 669 Mutex::Autolock _l(mLock); 670 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 671 mMasterMute = muted; 672 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 673 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 674 675 return NO_ERROR; 676} 677 678float AudioFlinger::masterVolume() const 679{ 680 Mutex::Autolock _l(mLock); 681 return masterVolume_l(); 682} 683 684float AudioFlinger::masterVolumeSW() const 685{ 686 Mutex::Autolock _l(mLock); 687 return masterVolumeSW_l(); 688} 689 690bool AudioFlinger::masterMute() const 691{ 692 Mutex::Autolock _l(mLock); 693 return masterMute_l(); 694} 695 696float AudioFlinger::masterVolume_l() const 697{ 698 if (MVS_FULL == mMasterVolumeSupportLvl) { 699 float ret_val; 700 AutoMutex lock(mHardwareLock); 701 702 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 703 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 704 (NULL != mPrimaryHardwareDev->get_master_volume), 705 "can't get master volume"); 706 707 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 708 mHardwareStatus = AUDIO_HW_IDLE; 709 return ret_val; 710 } 711 712 return mMasterVolume; 713} 714 715status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 716 audio_io_handle_t output) 717{ 718 // check calling permissions 719 if (!settingsAllowed()) { 720 return PERMISSION_DENIED; 721 } 722 723 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 724 ALOGE("setStreamVolume() invalid stream %d", stream); 725 return BAD_VALUE; 726 } 727 728 AutoMutex lock(mLock); 729 PlaybackThread *thread = NULL; 730 if (output) { 731 thread = checkPlaybackThread_l(output); 732 if (thread == NULL) { 733 return BAD_VALUE; 734 } 735 } 736 737 mStreamTypes[stream].volume = value; 738 739 if (thread == NULL) { 740 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 741 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 742 } 743 } else { 744 thread->setStreamVolume(stream, value); 745 } 746 747 return NO_ERROR; 748} 749 750status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 751{ 752 // check calling permissions 753 if (!settingsAllowed()) { 754 return PERMISSION_DENIED; 755 } 756 757 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 758 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 759 ALOGE("setStreamMute() invalid stream %d", stream); 760 return BAD_VALUE; 761 } 762 763 AutoMutex lock(mLock); 764 mStreamTypes[stream].mute = muted; 765 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 766 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 767 768 return NO_ERROR; 769} 770 771float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 772{ 773 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 774 return 0.0f; 775 } 776 777 AutoMutex lock(mLock); 778 float volume; 779 if (output) { 780 PlaybackThread *thread = checkPlaybackThread_l(output); 781 if (thread == NULL) { 782 return 0.0f; 783 } 784 volume = thread->streamVolume(stream); 785 } else { 786 volume = streamVolume_l(stream); 787 } 788 789 return volume; 790} 791 792bool AudioFlinger::streamMute(audio_stream_type_t stream) const 793{ 794 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 795 return true; 796 } 797 798 AutoMutex lock(mLock); 799 return streamMute_l(stream); 800} 801 802status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 803{ 804 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 805 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 806 // check calling permissions 807 if (!settingsAllowed()) { 808 return PERMISSION_DENIED; 809 } 810 811 // ioHandle == 0 means the parameters are global to the audio hardware interface 812 if (ioHandle == 0) { 813 Mutex::Autolock _l(mLock); 814 status_t final_result = NO_ERROR; 815 { 816 AutoMutex lock(mHardwareLock); 817 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 818 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 819 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 820 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 821 final_result = result ?: final_result; 822 } 823 mHardwareStatus = AUDIO_HW_IDLE; 824 } 825 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 826 AudioParameter param = AudioParameter(keyValuePairs); 827 String8 value; 828 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 829 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 830 if (mBtNrecIsOff != btNrecIsOff) { 831 for (size_t i = 0; i < mRecordThreads.size(); i++) { 832 sp<RecordThread> thread = mRecordThreads.valueAt(i); 833 RecordThread::RecordTrack *track = thread->track(); 834 if (track != NULL) { 835 audio_devices_t device = (audio_devices_t)( 836 thread->device() & AUDIO_DEVICE_IN_ALL); 837 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 838 thread->setEffectSuspended(FX_IID_AEC, 839 suspend, 840 track->sessionId()); 841 thread->setEffectSuspended(FX_IID_NS, 842 suspend, 843 track->sessionId()); 844 } 845 } 846 mBtNrecIsOff = btNrecIsOff; 847 } 848 } 849 return final_result; 850 } 851 852 // hold a strong ref on thread in case closeOutput() or closeInput() is called 853 // and the thread is exited once the lock is released 854 sp<ThreadBase> thread; 855 { 856 Mutex::Autolock _l(mLock); 857 thread = checkPlaybackThread_l(ioHandle); 858 if (thread == NULL) { 859 thread = checkRecordThread_l(ioHandle); 860 } else if (thread == primaryPlaybackThread_l()) { 861 // indicate output device change to all input threads for pre processing 862 AudioParameter param = AudioParameter(keyValuePairs); 863 int value; 864 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 865 (value != 0)) { 866 for (size_t i = 0; i < mRecordThreads.size(); i++) { 867 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 868 } 869 } 870 } 871 } 872 if (thread != 0) { 873 return thread->setParameters(keyValuePairs); 874 } 875 return BAD_VALUE; 876} 877 878String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 879{ 880// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 881// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 882 883 Mutex::Autolock _l(mLock); 884 885 if (ioHandle == 0) { 886 String8 out_s8; 887 888 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 889 char *s; 890 { 891 AutoMutex lock(mHardwareLock); 892 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 893 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 894 s = dev->get_parameters(dev, keys.string()); 895 mHardwareStatus = AUDIO_HW_IDLE; 896 } 897 out_s8 += String8(s ? s : ""); 898 free(s); 899 } 900 return out_s8; 901 } 902 903 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 904 if (playbackThread != NULL) { 905 return playbackThread->getParameters(keys); 906 } 907 RecordThread *recordThread = checkRecordThread_l(ioHandle); 908 if (recordThread != NULL) { 909 return recordThread->getParameters(keys); 910 } 911 return String8(""); 912} 913 914size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 915{ 916 status_t ret = initCheck(); 917 if (ret != NO_ERROR) { 918 return 0; 919 } 920 921 AutoMutex lock(mHardwareLock); 922 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 923 struct audio_config config = { 924 sample_rate: sampleRate, 925 channel_mask: audio_channel_in_mask_from_count(channelCount), 926 format: format, 927 }; 928 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 929 mHardwareStatus = AUDIO_HW_IDLE; 930 return size; 931} 932 933unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 934{ 935 if (ioHandle == 0) { 936 return 0; 937 } 938 939 Mutex::Autolock _l(mLock); 940 941 RecordThread *recordThread = checkRecordThread_l(ioHandle); 942 if (recordThread != NULL) { 943 return recordThread->getInputFramesLost(); 944 } 945 return 0; 946} 947 948status_t AudioFlinger::setVoiceVolume(float value) 949{ 950 status_t ret = initCheck(); 951 if (ret != NO_ERROR) { 952 return ret; 953 } 954 955 // check calling permissions 956 if (!settingsAllowed()) { 957 return PERMISSION_DENIED; 958 } 959 960 AutoMutex lock(mHardwareLock); 961 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 962 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 963 mHardwareStatus = AUDIO_HW_IDLE; 964 965 return ret; 966} 967 968status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 969 audio_io_handle_t output) const 970{ 971 status_t status; 972 973 Mutex::Autolock _l(mLock); 974 975 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 976 if (playbackThread != NULL) { 977 return playbackThread->getRenderPosition(halFrames, dspFrames); 978 } 979 980 return BAD_VALUE; 981} 982 983void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 984{ 985 986 Mutex::Autolock _l(mLock); 987 988 pid_t pid = IPCThreadState::self()->getCallingPid(); 989 if (mNotificationClients.indexOfKey(pid) < 0) { 990 sp<NotificationClient> notificationClient = new NotificationClient(this, 991 client, 992 pid); 993 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 994 995 mNotificationClients.add(pid, notificationClient); 996 997 sp<IBinder> binder = client->asBinder(); 998 binder->linkToDeath(notificationClient); 999 1000 // the config change is always sent from playback or record threads to avoid deadlock 1001 // with AudioSystem::gLock 1002 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1003 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1004 } 1005 1006 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1007 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1008 } 1009 } 1010} 1011 1012void AudioFlinger::removeNotificationClient(pid_t pid) 1013{ 1014 Mutex::Autolock _l(mLock); 1015 1016 mNotificationClients.removeItem(pid); 1017 1018 ALOGV("%d died, releasing its sessions", pid); 1019 size_t num = mAudioSessionRefs.size(); 1020 bool removed = false; 1021 for (size_t i = 0; i< num; ) { 1022 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1023 ALOGV(" pid %d @ %d", ref->mPid, i); 1024 if (ref->mPid == pid) { 1025 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1026 mAudioSessionRefs.removeAt(i); 1027 delete ref; 1028 removed = true; 1029 num--; 1030 } else { 1031 i++; 1032 } 1033 } 1034 if (removed) { 1035 purgeStaleEffects_l(); 1036 } 1037} 1038 1039// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1040void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1041{ 1042 size_t size = mNotificationClients.size(); 1043 for (size_t i = 0; i < size; i++) { 1044 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1045 param2); 1046 } 1047} 1048 1049// removeClient_l() must be called with AudioFlinger::mLock held 1050void AudioFlinger::removeClient_l(pid_t pid) 1051{ 1052 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1053 mClients.removeItem(pid); 1054} 1055 1056 1057// ---------------------------------------------------------------------------- 1058 1059AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1060 uint32_t device, type_t type) 1061 : Thread(false), 1062 mType(type), 1063 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1064 // mChannelMask 1065 mChannelCount(0), 1066 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1067 mParamStatus(NO_ERROR), 1068 mStandby(false), mId(id), 1069 mDevice(device), 1070 mDeathRecipient(new PMDeathRecipient(this)) 1071{ 1072} 1073 1074AudioFlinger::ThreadBase::~ThreadBase() 1075{ 1076 mParamCond.broadcast(); 1077 // do not lock the mutex in destructor 1078 releaseWakeLock_l(); 1079 if (mPowerManager != 0) { 1080 sp<IBinder> binder = mPowerManager->asBinder(); 1081 binder->unlinkToDeath(mDeathRecipient); 1082 } 1083} 1084 1085void AudioFlinger::ThreadBase::exit() 1086{ 1087 ALOGV("ThreadBase::exit"); 1088 { 1089 // This lock prevents the following race in thread (uniprocessor for illustration): 1090 // if (!exitPending()) { 1091 // // context switch from here to exit() 1092 // // exit() calls requestExit(), what exitPending() observes 1093 // // exit() calls signal(), which is dropped since no waiters 1094 // // context switch back from exit() to here 1095 // mWaitWorkCV.wait(...); 1096 // // now thread is hung 1097 // } 1098 AutoMutex lock(mLock); 1099 requestExit(); 1100 mWaitWorkCV.signal(); 1101 } 1102 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1103 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1104 requestExitAndWait(); 1105} 1106 1107status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1108{ 1109 status_t status; 1110 1111 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1112 Mutex::Autolock _l(mLock); 1113 1114 mNewParameters.add(keyValuePairs); 1115 mWaitWorkCV.signal(); 1116 // wait condition with timeout in case the thread loop has exited 1117 // before the request could be processed 1118 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1119 status = mParamStatus; 1120 mWaitWorkCV.signal(); 1121 } else { 1122 status = TIMED_OUT; 1123 } 1124 return status; 1125} 1126 1127void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1128{ 1129 Mutex::Autolock _l(mLock); 1130 sendConfigEvent_l(event, param); 1131} 1132 1133// sendConfigEvent_l() must be called with ThreadBase::mLock held 1134void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1135{ 1136 ConfigEvent configEvent; 1137 configEvent.mEvent = event; 1138 configEvent.mParam = param; 1139 mConfigEvents.add(configEvent); 1140 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1141 mWaitWorkCV.signal(); 1142} 1143 1144void AudioFlinger::ThreadBase::processConfigEvents() 1145{ 1146 mLock.lock(); 1147 while (!mConfigEvents.isEmpty()) { 1148 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1149 ConfigEvent configEvent = mConfigEvents[0]; 1150 mConfigEvents.removeAt(0); 1151 // release mLock before locking AudioFlinger mLock: lock order is always 1152 // AudioFlinger then ThreadBase to avoid cross deadlock 1153 mLock.unlock(); 1154 mAudioFlinger->mLock.lock(); 1155 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1156 mAudioFlinger->mLock.unlock(); 1157 mLock.lock(); 1158 } 1159 mLock.unlock(); 1160} 1161 1162status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1163{ 1164 const size_t SIZE = 256; 1165 char buffer[SIZE]; 1166 String8 result; 1167 1168 bool locked = tryLock(mLock); 1169 if (!locked) { 1170 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1171 write(fd, buffer, strlen(buffer)); 1172 } 1173 1174 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1175 result.append(buffer); 1176 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1177 result.append(buffer); 1178 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1179 result.append(buffer); 1180 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1181 result.append(buffer); 1182 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1183 result.append(buffer); 1184 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1185 result.append(buffer); 1186 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1187 result.append(buffer); 1188 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1189 result.append(buffer); 1190 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1191 result.append(buffer); 1192 1193 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1194 result.append(buffer); 1195 result.append(" Index Command"); 1196 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1197 snprintf(buffer, SIZE, "\n %02d ", i); 1198 result.append(buffer); 1199 result.append(mNewParameters[i]); 1200 } 1201 1202 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1203 result.append(buffer); 1204 snprintf(buffer, SIZE, " Index event param\n"); 1205 result.append(buffer); 1206 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1207 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1208 result.append(buffer); 1209 } 1210 result.append("\n"); 1211 1212 write(fd, result.string(), result.size()); 1213 1214 if (locked) { 1215 mLock.unlock(); 1216 } 1217 return NO_ERROR; 1218} 1219 1220status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1221{ 1222 const size_t SIZE = 256; 1223 char buffer[SIZE]; 1224 String8 result; 1225 1226 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1227 write(fd, buffer, strlen(buffer)); 1228 1229 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1230 sp<EffectChain> chain = mEffectChains[i]; 1231 if (chain != 0) { 1232 chain->dump(fd, args); 1233 } 1234 } 1235 return NO_ERROR; 1236} 1237 1238void AudioFlinger::ThreadBase::acquireWakeLock() 1239{ 1240 Mutex::Autolock _l(mLock); 1241 acquireWakeLock_l(); 1242} 1243 1244void AudioFlinger::ThreadBase::acquireWakeLock_l() 1245{ 1246 if (mPowerManager == 0) { 1247 // use checkService() to avoid blocking if power service is not up yet 1248 sp<IBinder> binder = 1249 defaultServiceManager()->checkService(String16("power")); 1250 if (binder == 0) { 1251 ALOGW("Thread %s cannot connect to the power manager service", mName); 1252 } else { 1253 mPowerManager = interface_cast<IPowerManager>(binder); 1254 binder->linkToDeath(mDeathRecipient); 1255 } 1256 } 1257 if (mPowerManager != 0) { 1258 sp<IBinder> binder = new BBinder(); 1259 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1260 binder, 1261 String16(mName)); 1262 if (status == NO_ERROR) { 1263 mWakeLockToken = binder; 1264 } 1265 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1266 } 1267} 1268 1269void AudioFlinger::ThreadBase::releaseWakeLock() 1270{ 1271 Mutex::Autolock _l(mLock); 1272 releaseWakeLock_l(); 1273} 1274 1275void AudioFlinger::ThreadBase::releaseWakeLock_l() 1276{ 1277 if (mWakeLockToken != 0) { 1278 ALOGV("releaseWakeLock_l() %s", mName); 1279 if (mPowerManager != 0) { 1280 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1281 } 1282 mWakeLockToken.clear(); 1283 } 1284} 1285 1286void AudioFlinger::ThreadBase::clearPowerManager() 1287{ 1288 Mutex::Autolock _l(mLock); 1289 releaseWakeLock_l(); 1290 mPowerManager.clear(); 1291} 1292 1293void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1294{ 1295 sp<ThreadBase> thread = mThread.promote(); 1296 if (thread != 0) { 1297 thread->clearPowerManager(); 1298 } 1299 ALOGW("power manager service died !!!"); 1300} 1301 1302void AudioFlinger::ThreadBase::setEffectSuspended( 1303 const effect_uuid_t *type, bool suspend, int sessionId) 1304{ 1305 Mutex::Autolock _l(mLock); 1306 setEffectSuspended_l(type, suspend, sessionId); 1307} 1308 1309void AudioFlinger::ThreadBase::setEffectSuspended_l( 1310 const effect_uuid_t *type, bool suspend, int sessionId) 1311{ 1312 sp<EffectChain> chain = getEffectChain_l(sessionId); 1313 if (chain != 0) { 1314 if (type != NULL) { 1315 chain->setEffectSuspended_l(type, suspend); 1316 } else { 1317 chain->setEffectSuspendedAll_l(suspend); 1318 } 1319 } 1320 1321 updateSuspendedSessions_l(type, suspend, sessionId); 1322} 1323 1324void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1325{ 1326 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1327 if (index < 0) { 1328 return; 1329 } 1330 1331 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1332 mSuspendedSessions.editValueAt(index); 1333 1334 for (size_t i = 0; i < sessionEffects.size(); i++) { 1335 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1336 for (int j = 0; j < desc->mRefCount; j++) { 1337 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1338 chain->setEffectSuspendedAll_l(true); 1339 } else { 1340 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1341 desc->mType.timeLow); 1342 chain->setEffectSuspended_l(&desc->mType, true); 1343 } 1344 } 1345 } 1346} 1347 1348void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1349 bool suspend, 1350 int sessionId) 1351{ 1352 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1353 1354 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1355 1356 if (suspend) { 1357 if (index >= 0) { 1358 sessionEffects = mSuspendedSessions.editValueAt(index); 1359 } else { 1360 mSuspendedSessions.add(sessionId, sessionEffects); 1361 } 1362 } else { 1363 if (index < 0) { 1364 return; 1365 } 1366 sessionEffects = mSuspendedSessions.editValueAt(index); 1367 } 1368 1369 1370 int key = EffectChain::kKeyForSuspendAll; 1371 if (type != NULL) { 1372 key = type->timeLow; 1373 } 1374 index = sessionEffects.indexOfKey(key); 1375 1376 sp<SuspendedSessionDesc> desc; 1377 if (suspend) { 1378 if (index >= 0) { 1379 desc = sessionEffects.valueAt(index); 1380 } else { 1381 desc = new SuspendedSessionDesc(); 1382 if (type != NULL) { 1383 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1384 } 1385 sessionEffects.add(key, desc); 1386 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1387 } 1388 desc->mRefCount++; 1389 } else { 1390 if (index < 0) { 1391 return; 1392 } 1393 desc = sessionEffects.valueAt(index); 1394 if (--desc->mRefCount == 0) { 1395 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1396 sessionEffects.removeItemsAt(index); 1397 if (sessionEffects.isEmpty()) { 1398 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1399 sessionId); 1400 mSuspendedSessions.removeItem(sessionId); 1401 } 1402 } 1403 } 1404 if (!sessionEffects.isEmpty()) { 1405 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1406 } 1407} 1408 1409void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1410 bool enabled, 1411 int sessionId) 1412{ 1413 Mutex::Autolock _l(mLock); 1414 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1415} 1416 1417void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1418 bool enabled, 1419 int sessionId) 1420{ 1421 if (mType != RECORD) { 1422 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1423 // another session. This gives the priority to well behaved effect control panels 1424 // and applications not using global effects. 1425 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1426 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1427 } 1428 } 1429 1430 sp<EffectChain> chain = getEffectChain_l(sessionId); 1431 if (chain != 0) { 1432 chain->checkSuspendOnEffectEnabled(effect, enabled); 1433 } 1434} 1435 1436// ---------------------------------------------------------------------------- 1437 1438AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1439 AudioStreamOut* output, 1440 audio_io_handle_t id, 1441 uint32_t device, 1442 type_t type) 1443 : ThreadBase(audioFlinger, id, device, type), 1444 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1445 // Assumes constructor is called by AudioFlinger with it's mLock held, 1446 // but it would be safer to explicitly pass initial masterMute as parameter 1447 mMasterMute(audioFlinger->masterMute_l()), 1448 // mStreamTypes[] initialized in constructor body 1449 mOutput(output), 1450 // Assumes constructor is called by AudioFlinger with it's mLock held, 1451 // but it would be safer to explicitly pass initial masterVolume as parameter 1452 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1453 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1454 mMixerStatus(MIXER_IDLE), 1455 mPrevMixerStatus(MIXER_IDLE), 1456 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1457{ 1458 snprintf(mName, kNameLength, "AudioOut_%X", id); 1459 1460 readOutputParameters(); 1461 1462 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1463 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1464 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1465 stream = (audio_stream_type_t) (stream + 1)) { 1466 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1467 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1468 } 1469 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1470 // because mAudioFlinger doesn't have one to copy from 1471} 1472 1473AudioFlinger::PlaybackThread::~PlaybackThread() 1474{ 1475 delete [] mMixBuffer; 1476} 1477 1478status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1479{ 1480 dumpInternals(fd, args); 1481 dumpTracks(fd, args); 1482 dumpEffectChains(fd, args); 1483 return NO_ERROR; 1484} 1485 1486status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1487{ 1488 const size_t SIZE = 256; 1489 char buffer[SIZE]; 1490 String8 result; 1491 1492 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1493 result.append(buffer); 1494 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1495 for (size_t i = 0; i < mTracks.size(); ++i) { 1496 sp<Track> track = mTracks[i]; 1497 if (track != 0) { 1498 track->dump(buffer, SIZE); 1499 result.append(buffer); 1500 } 1501 } 1502 1503 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1504 result.append(buffer); 1505 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1506 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1507 sp<Track> track = mActiveTracks[i].promote(); 1508 if (track != 0) { 1509 track->dump(buffer, SIZE); 1510 result.append(buffer); 1511 } 1512 } 1513 write(fd, result.string(), result.size()); 1514 return NO_ERROR; 1515} 1516 1517status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1518{ 1519 const size_t SIZE = 256; 1520 char buffer[SIZE]; 1521 String8 result; 1522 1523 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1524 result.append(buffer); 1525 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1526 result.append(buffer); 1527 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1528 result.append(buffer); 1529 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1530 result.append(buffer); 1531 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1532 result.append(buffer); 1533 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1534 result.append(buffer); 1535 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1536 result.append(buffer); 1537 write(fd, result.string(), result.size()); 1538 1539 dumpBase(fd, args); 1540 1541 return NO_ERROR; 1542} 1543 1544// Thread virtuals 1545status_t AudioFlinger::PlaybackThread::readyToRun() 1546{ 1547 status_t status = initCheck(); 1548 if (status == NO_ERROR) { 1549 ALOGI("AudioFlinger's thread %p ready to run", this); 1550 } else { 1551 ALOGE("No working audio driver found."); 1552 } 1553 return status; 1554} 1555 1556void AudioFlinger::PlaybackThread::onFirstRef() 1557{ 1558 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1559} 1560 1561// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1562sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1563 const sp<AudioFlinger::Client>& client, 1564 audio_stream_type_t streamType, 1565 uint32_t sampleRate, 1566 audio_format_t format, 1567 uint32_t channelMask, 1568 int frameCount, 1569 const sp<IMemory>& sharedBuffer, 1570 int sessionId, 1571 IAudioFlinger::track_flags_t flags, 1572 pid_t tid, 1573 status_t *status) 1574{ 1575 sp<Track> track; 1576 status_t lStatus; 1577 1578 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1579 1580 // client expresses a preference for FAST, but we get the final say 1581 if ((flags & IAudioFlinger::TRACK_FAST) && 1582 !( 1583 // not timed 1584 (!isTimed) && 1585 // either of these use cases: 1586 ( 1587 // use case 1: shared buffer with any frame count 1588 ( 1589 (sharedBuffer != 0) 1590 ) || 1591 // use case 2: callback handler and small power-of-2 frame count 1592 ( 1593 (tid != -1) && 1594 // FIXME supported frame counts should not be hard-coded 1595 ( 1596 (frameCount == 128) || 1597 (frameCount == 256) || 1598 (frameCount == 512) 1599 ) 1600 ) 1601 ) && 1602 // PCM data 1603 audio_is_linear_pcm(format) && 1604 // mono or stereo 1605 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1606 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1607 // hardware sample rate 1608 (sampleRate == mSampleRate) 1609 // FIXME test that MixerThread for this fast track has a capable output HAL 1610 // FIXME add a permission test also? 1611 ) ) { 1612 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied"); 1613 flags &= ~IAudioFlinger::TRACK_FAST; 1614 } 1615 1616 if (mType == DIRECT) { 1617 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1618 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1619 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1620 "for output %p with format %d", 1621 sampleRate, format, channelMask, mOutput, mFormat); 1622 lStatus = BAD_VALUE; 1623 goto Exit; 1624 } 1625 } 1626 } else { 1627 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1628 if (sampleRate > mSampleRate*2) { 1629 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1630 lStatus = BAD_VALUE; 1631 goto Exit; 1632 } 1633 } 1634 1635 lStatus = initCheck(); 1636 if (lStatus != NO_ERROR) { 1637 ALOGE("Audio driver not initialized."); 1638 goto Exit; 1639 } 1640 1641 { // scope for mLock 1642 Mutex::Autolock _l(mLock); 1643 1644 // all tracks in same audio session must share the same routing strategy otherwise 1645 // conflicts will happen when tracks are moved from one output to another by audio policy 1646 // manager 1647 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1648 for (size_t i = 0; i < mTracks.size(); ++i) { 1649 sp<Track> t = mTracks[i]; 1650 if (t != 0 && !t->isOutputTrack()) { 1651 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1652 if (sessionId == t->sessionId() && strategy != actual) { 1653 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1654 strategy, actual); 1655 lStatus = BAD_VALUE; 1656 goto Exit; 1657 } 1658 } 1659 } 1660 1661 if (!isTimed) { 1662 track = new Track(this, client, streamType, sampleRate, format, 1663 channelMask, frameCount, sharedBuffer, sessionId, flags); 1664 } else { 1665 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1666 channelMask, frameCount, sharedBuffer, sessionId); 1667 } 1668 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1669 lStatus = NO_MEMORY; 1670 goto Exit; 1671 } 1672 mTracks.add(track); 1673 1674 sp<EffectChain> chain = getEffectChain_l(sessionId); 1675 if (chain != 0) { 1676 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1677 track->setMainBuffer(chain->inBuffer()); 1678 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1679 chain->incTrackCnt(); 1680 } 1681 } 1682 1683#ifdef HAVE_REQUEST_PRIORITY 1684 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1685 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1686 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1687 // so ask activity manager to do this on our behalf 1688 int err = requestPriority(callingPid, tid, 1); 1689 if (err != 0) { 1690 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1691 1, callingPid, tid, err); 1692 } 1693 } 1694#endif 1695 1696 lStatus = NO_ERROR; 1697 1698Exit: 1699 if (status) { 1700 *status = lStatus; 1701 } 1702 return track; 1703} 1704 1705uint32_t AudioFlinger::PlaybackThread::latency() const 1706{ 1707 Mutex::Autolock _l(mLock); 1708 if (initCheck() == NO_ERROR) { 1709 return mOutput->stream->get_latency(mOutput->stream); 1710 } else { 1711 return 0; 1712 } 1713} 1714 1715void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1716{ 1717 Mutex::Autolock _l(mLock); 1718 mMasterVolume = value; 1719} 1720 1721void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1722{ 1723 Mutex::Autolock _l(mLock); 1724 setMasterMute_l(muted); 1725} 1726 1727void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1728{ 1729 Mutex::Autolock _l(mLock); 1730 mStreamTypes[stream].volume = value; 1731} 1732 1733void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1734{ 1735 Mutex::Autolock _l(mLock); 1736 mStreamTypes[stream].mute = muted; 1737} 1738 1739float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1740{ 1741 Mutex::Autolock _l(mLock); 1742 return mStreamTypes[stream].volume; 1743} 1744 1745// addTrack_l() must be called with ThreadBase::mLock held 1746status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1747{ 1748 status_t status = ALREADY_EXISTS; 1749 1750 // set retry count for buffer fill 1751 track->mRetryCount = kMaxTrackStartupRetries; 1752 if (mActiveTracks.indexOf(track) < 0) { 1753 // the track is newly added, make sure it fills up all its 1754 // buffers before playing. This is to ensure the client will 1755 // effectively get the latency it requested. 1756 track->mFillingUpStatus = Track::FS_FILLING; 1757 track->mResetDone = false; 1758 mActiveTracks.add(track); 1759 if (track->mainBuffer() != mMixBuffer) { 1760 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1761 if (chain != 0) { 1762 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1763 chain->incActiveTrackCnt(); 1764 } 1765 } 1766 1767 status = NO_ERROR; 1768 } 1769 1770 ALOGV("mWaitWorkCV.broadcast"); 1771 mWaitWorkCV.broadcast(); 1772 1773 return status; 1774} 1775 1776// destroyTrack_l() must be called with ThreadBase::mLock held 1777void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1778{ 1779 track->mState = TrackBase::TERMINATED; 1780 if (mActiveTracks.indexOf(track) < 0) { 1781 removeTrack_l(track); 1782 } 1783} 1784 1785void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1786{ 1787 mTracks.remove(track); 1788 deleteTrackName_l(track->name()); 1789 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1790 if (chain != 0) { 1791 chain->decTrackCnt(); 1792 } 1793} 1794 1795String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1796{ 1797 String8 out_s8 = String8(""); 1798 char *s; 1799 1800 Mutex::Autolock _l(mLock); 1801 if (initCheck() != NO_ERROR) { 1802 return out_s8; 1803 } 1804 1805 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1806 out_s8 = String8(s); 1807 free(s); 1808 return out_s8; 1809} 1810 1811// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1812void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1813 AudioSystem::OutputDescriptor desc; 1814 void *param2 = NULL; 1815 1816 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1817 1818 switch (event) { 1819 case AudioSystem::OUTPUT_OPENED: 1820 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1821 desc.channels = mChannelMask; 1822 desc.samplingRate = mSampleRate; 1823 desc.format = mFormat; 1824 desc.frameCount = mFrameCount; 1825 desc.latency = latency(); 1826 param2 = &desc; 1827 break; 1828 1829 case AudioSystem::STREAM_CONFIG_CHANGED: 1830 param2 = ¶m; 1831 case AudioSystem::OUTPUT_CLOSED: 1832 default: 1833 break; 1834 } 1835 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1836} 1837 1838void AudioFlinger::PlaybackThread::readOutputParameters() 1839{ 1840 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1841 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1842 mChannelCount = (uint16_t)popcount(mChannelMask); 1843 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1844 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1845 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1846 1847 // FIXME - Current mixer implementation only supports stereo output: Always 1848 // Allocate a stereo buffer even if HW output is mono. 1849 delete[] mMixBuffer; 1850 mMixBuffer = new int16_t[mFrameCount * 2]; 1851 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1852 1853 // force reconfiguration of effect chains and engines to take new buffer size and audio 1854 // parameters into account 1855 // Note that mLock is not held when readOutputParameters() is called from the constructor 1856 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1857 // matter. 1858 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1859 Vector< sp<EffectChain> > effectChains = mEffectChains; 1860 for (size_t i = 0; i < effectChains.size(); i ++) { 1861 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1862 } 1863} 1864 1865status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1866{ 1867 if (halFrames == NULL || dspFrames == NULL) { 1868 return BAD_VALUE; 1869 } 1870 Mutex::Autolock _l(mLock); 1871 if (initCheck() != NO_ERROR) { 1872 return INVALID_OPERATION; 1873 } 1874 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1875 1876 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1877} 1878 1879uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1880{ 1881 Mutex::Autolock _l(mLock); 1882 uint32_t result = 0; 1883 if (getEffectChain_l(sessionId) != 0) { 1884 result = EFFECT_SESSION; 1885 } 1886 1887 for (size_t i = 0; i < mTracks.size(); ++i) { 1888 sp<Track> track = mTracks[i]; 1889 if (sessionId == track->sessionId() && 1890 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1891 result |= TRACK_SESSION; 1892 break; 1893 } 1894 } 1895 1896 return result; 1897} 1898 1899uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1900{ 1901 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1902 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1903 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1904 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1905 } 1906 for (size_t i = 0; i < mTracks.size(); i++) { 1907 sp<Track> track = mTracks[i]; 1908 if (sessionId == track->sessionId() && 1909 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1910 return AudioSystem::getStrategyForStream(track->streamType()); 1911 } 1912 } 1913 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1914} 1915 1916 1917AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1918{ 1919 Mutex::Autolock _l(mLock); 1920 return mOutput; 1921} 1922 1923AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1924{ 1925 Mutex::Autolock _l(mLock); 1926 AudioStreamOut *output = mOutput; 1927 mOutput = NULL; 1928 return output; 1929} 1930 1931// this method must always be called either with ThreadBase mLock held or inside the thread loop 1932audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1933{ 1934 if (mOutput == NULL) { 1935 return NULL; 1936 } 1937 return &mOutput->stream->common; 1938} 1939 1940uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1941{ 1942 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1943 // decoding and transfer time. So sleeping for half of the latency would likely cause 1944 // underruns 1945 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1946 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1947 } else { 1948 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1949 } 1950} 1951 1952status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1953{ 1954 if (!isValidSyncEvent(event)) { 1955 return BAD_VALUE; 1956 } 1957 1958 Mutex::Autolock _l(mLock); 1959 1960 for (size_t i = 0; i < mTracks.size(); ++i) { 1961 sp<Track> track = mTracks[i]; 1962 if (event->triggerSession() == track->sessionId()) { 1963 track->setSyncEvent(event); 1964 return NO_ERROR; 1965 } 1966 } 1967 1968 return NAME_NOT_FOUND; 1969} 1970 1971bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 1972{ 1973 switch (event->type()) { 1974 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 1975 return true; 1976 default: 1977 break; 1978 } 1979 return false; 1980} 1981 1982// ---------------------------------------------------------------------------- 1983 1984AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1985 audio_io_handle_t id, uint32_t device, type_t type) 1986 : PlaybackThread(audioFlinger, output, id, device, type) 1987{ 1988 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1989 // FIXME - Current mixer implementation only supports stereo output 1990 if (mChannelCount == 1) { 1991 ALOGE("Invalid audio hardware channel count"); 1992 } 1993} 1994 1995AudioFlinger::MixerThread::~MixerThread() 1996{ 1997 delete mAudioMixer; 1998} 1999 2000class CpuStats { 2001public: 2002 CpuStats(); 2003 void sample(const String8 &title); 2004#ifdef DEBUG_CPU_USAGE 2005private: 2006 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2007 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2008 2009 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2010 2011 int mCpuNum; // thread's current CPU number 2012 int mCpukHz; // frequency of thread's current CPU in kHz 2013#endif 2014}; 2015 2016CpuStats::CpuStats() 2017#ifdef DEBUG_CPU_USAGE 2018 : mCpuNum(-1), mCpukHz(-1) 2019#endif 2020{ 2021} 2022 2023void CpuStats::sample(const String8 &title) { 2024#ifdef DEBUG_CPU_USAGE 2025 // get current thread's delta CPU time in wall clock ns 2026 double wcNs; 2027 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2028 2029 // record sample for wall clock statistics 2030 if (valid) { 2031 mWcStats.sample(wcNs); 2032 } 2033 2034 // get the current CPU number 2035 int cpuNum = sched_getcpu(); 2036 2037 // get the current CPU frequency in kHz 2038 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2039 2040 // check if either CPU number or frequency changed 2041 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2042 mCpuNum = cpuNum; 2043 mCpukHz = cpukHz; 2044 // ignore sample for purposes of cycles 2045 valid = false; 2046 } 2047 2048 // if no change in CPU number or frequency, then record sample for cycle statistics 2049 if (valid && mCpukHz > 0) { 2050 double cycles = wcNs * cpukHz * 0.000001; 2051 mHzStats.sample(cycles); 2052 } 2053 2054 unsigned n = mWcStats.n(); 2055 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2056 if ((n & 127) == 1) { 2057 long long elapsed = mCpuUsage.elapsed(); 2058 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2059 double perLoop = elapsed / (double) n; 2060 double perLoop100 = perLoop * 0.01; 2061 double perLoop1k = perLoop * 0.001; 2062 double mean = mWcStats.mean(); 2063 double stddev = mWcStats.stddev(); 2064 double minimum = mWcStats.minimum(); 2065 double maximum = mWcStats.maximum(); 2066 double meanCycles = mHzStats.mean(); 2067 double stddevCycles = mHzStats.stddev(); 2068 double minCycles = mHzStats.minimum(); 2069 double maxCycles = mHzStats.maximum(); 2070 mCpuUsage.resetElapsed(); 2071 mWcStats.reset(); 2072 mHzStats.reset(); 2073 ALOGD("CPU usage for %s over past %.1f secs\n" 2074 " (%u mixer loops at %.1f mean ms per loop):\n" 2075 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2076 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2077 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2078 title.string(), 2079 elapsed * .000000001, n, perLoop * .000001, 2080 mean * .001, 2081 stddev * .001, 2082 minimum * .001, 2083 maximum * .001, 2084 mean / perLoop100, 2085 stddev / perLoop100, 2086 minimum / perLoop100, 2087 maximum / perLoop100, 2088 meanCycles / perLoop1k, 2089 stddevCycles / perLoop1k, 2090 minCycles / perLoop1k, 2091 maxCycles / perLoop1k); 2092 2093 } 2094 } 2095#endif 2096}; 2097 2098void AudioFlinger::PlaybackThread::checkSilentMode_l() 2099{ 2100 if (!mMasterMute) { 2101 char value[PROPERTY_VALUE_MAX]; 2102 if (property_get("ro.audio.silent", value, "0") > 0) { 2103 char *endptr; 2104 unsigned long ul = strtoul(value, &endptr, 0); 2105 if (*endptr == '\0' && ul != 0) { 2106 ALOGD("Silence is golden"); 2107 // The setprop command will not allow a property to be changed after 2108 // the first time it is set, so we don't have to worry about un-muting. 2109 setMasterMute_l(true); 2110 } 2111 } 2112 } 2113} 2114 2115bool AudioFlinger::PlaybackThread::threadLoop() 2116{ 2117 Vector< sp<Track> > tracksToRemove; 2118 2119 standbyTime = systemTime(); 2120 2121 // MIXER 2122 nsecs_t lastWarning = 0; 2123if (mType == MIXER) { 2124 longStandbyExit = false; 2125} 2126 2127 // DUPLICATING 2128 // FIXME could this be made local to while loop? 2129 writeFrames = 0; 2130 2131 cacheParameters_l(); 2132 sleepTime = idleSleepTime; 2133 2134if (mType == MIXER) { 2135 sleepTimeShift = 0; 2136} 2137 2138 CpuStats cpuStats; 2139 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2140 2141 acquireWakeLock(); 2142 2143 while (!exitPending()) 2144 { 2145 cpuStats.sample(myName); 2146 2147 Vector< sp<EffectChain> > effectChains; 2148 2149 processConfigEvents(); 2150 2151 { // scope for mLock 2152 2153 Mutex::Autolock _l(mLock); 2154 2155 if (checkForNewParameters_l()) { 2156 cacheParameters_l(); 2157 } 2158 2159 saveOutputTracks(); 2160 2161 // put audio hardware into standby after short delay 2162 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2163 mSuspended > 0)) { 2164 if (!mStandby) { 2165 2166 threadLoop_standby(); 2167 2168 mStandby = true; 2169 mBytesWritten = 0; 2170 } 2171 2172 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2173 // we're about to wait, flush the binder command buffer 2174 IPCThreadState::self()->flushCommands(); 2175 2176 clearOutputTracks(); 2177 2178 if (exitPending()) break; 2179 2180 releaseWakeLock_l(); 2181 // wait until we have something to do... 2182 ALOGV("%s going to sleep", myName.string()); 2183 mWaitWorkCV.wait(mLock); 2184 ALOGV("%s waking up", myName.string()); 2185 acquireWakeLock_l(); 2186 2187 mPrevMixerStatus = MIXER_IDLE; 2188 2189 checkSilentMode_l(); 2190 2191 standbyTime = systemTime() + standbyDelay; 2192 sleepTime = idleSleepTime; 2193 if (mType == MIXER) { 2194 sleepTimeShift = 0; 2195 } 2196 2197 continue; 2198 } 2199 } 2200 2201 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2202 // Shift in the new status; this could be a queue if it's 2203 // useful to filter the mixer status over several cycles. 2204 mPrevMixerStatus = mMixerStatus; 2205 mMixerStatus = newMixerStatus; 2206 2207 // prevent any changes in effect chain list and in each effect chain 2208 // during mixing and effect process as the audio buffers could be deleted 2209 // or modified if an effect is created or deleted 2210 lockEffectChains_l(effectChains); 2211 } 2212 2213 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2214 threadLoop_mix(); 2215 } else { 2216 threadLoop_sleepTime(); 2217 } 2218 2219 if (mSuspended > 0) { 2220 sleepTime = suspendSleepTimeUs(); 2221 } 2222 2223 // only process effects if we're going to write 2224 if (sleepTime == 0) { 2225 for (size_t i = 0; i < effectChains.size(); i ++) { 2226 effectChains[i]->process_l(); 2227 } 2228 } 2229 2230 // enable changes in effect chain 2231 unlockEffectChains(effectChains); 2232 2233 // sleepTime == 0 means we must write to audio hardware 2234 if (sleepTime == 0) { 2235 2236 threadLoop_write(); 2237 2238if (mType == MIXER) { 2239 // write blocked detection 2240 nsecs_t now = systemTime(); 2241 nsecs_t delta = now - mLastWriteTime; 2242 if (!mStandby && delta > maxPeriod) { 2243 mNumDelayedWrites++; 2244 if ((now - lastWarning) > kWarningThrottleNs) { 2245 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2246 ns2ms(delta), mNumDelayedWrites, this); 2247 lastWarning = now; 2248 } 2249 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2250 // a different threshold. Or completely removed for what it is worth anyway... 2251 if (mStandby) { 2252 longStandbyExit = true; 2253 } 2254 } 2255} 2256 2257 mStandby = false; 2258 } else { 2259 usleep(sleepTime); 2260 } 2261 2262 // finally let go of removed track(s), without the lock held 2263 // since we can't guarantee the destructors won't acquire that 2264 // same lock. 2265 tracksToRemove.clear(); 2266 2267 // FIXME I don't understand the need for this here; 2268 // it was in the original code but maybe the 2269 // assignment in saveOutputTracks() makes this unnecessary? 2270 clearOutputTracks(); 2271 2272 // Effect chains will be actually deleted here if they were removed from 2273 // mEffectChains list during mixing or effects processing 2274 effectChains.clear(); 2275 2276 // FIXME Note that the above .clear() is no longer necessary since effectChains 2277 // is now local to this block, but will keep it for now (at least until merge done). 2278 } 2279 2280if (mType == MIXER || mType == DIRECT) { 2281 // put output stream into standby mode 2282 if (!mStandby) { 2283 mOutput->stream->common.standby(&mOutput->stream->common); 2284 } 2285} 2286if (mType == DUPLICATING) { 2287 // for DuplicatingThread, standby mode is handled by the outputTracks 2288} 2289 2290 releaseWakeLock(); 2291 2292 ALOGV("Thread %p type %d exiting", this, mType); 2293 return false; 2294} 2295 2296// shared by MIXER and DIRECT, overridden by DUPLICATING 2297void AudioFlinger::PlaybackThread::threadLoop_write() 2298{ 2299 // FIXME rewrite to reduce number of system calls 2300 mLastWriteTime = systemTime(); 2301 mInWrite = true; 2302 mBytesWritten += mixBufferSize; 2303 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2304 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2305 mNumWrites++; 2306 mInWrite = false; 2307} 2308 2309// shared by MIXER and DIRECT, overridden by DUPLICATING 2310void AudioFlinger::PlaybackThread::threadLoop_standby() 2311{ 2312 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2313 mOutput->stream->common.standby(&mOutput->stream->common); 2314} 2315 2316void AudioFlinger::MixerThread::threadLoop_mix() 2317{ 2318 // obtain the presentation timestamp of the next output buffer 2319 int64_t pts; 2320 status_t status = INVALID_OPERATION; 2321 2322 if (NULL != mOutput->stream->get_next_write_timestamp) { 2323 status = mOutput->stream->get_next_write_timestamp( 2324 mOutput->stream, &pts); 2325 } 2326 2327 if (status != NO_ERROR) { 2328 pts = AudioBufferProvider::kInvalidPTS; 2329 } 2330 2331 // mix buffers... 2332 mAudioMixer->process(pts); 2333 // increase sleep time progressively when application underrun condition clears. 2334 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2335 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2336 // such that we would underrun the audio HAL. 2337 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2338 sleepTimeShift--; 2339 } 2340 sleepTime = 0; 2341 standbyTime = systemTime() + standbyDelay; 2342 //TODO: delay standby when effects have a tail 2343} 2344 2345void AudioFlinger::MixerThread::threadLoop_sleepTime() 2346{ 2347 // If no tracks are ready, sleep once for the duration of an output 2348 // buffer size, then write 0s to the output 2349 if (sleepTime == 0) { 2350 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2351 sleepTime = activeSleepTime >> sleepTimeShift; 2352 if (sleepTime < kMinThreadSleepTimeUs) { 2353 sleepTime = kMinThreadSleepTimeUs; 2354 } 2355 // reduce sleep time in case of consecutive application underruns to avoid 2356 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2357 // duration we would end up writing less data than needed by the audio HAL if 2358 // the condition persists. 2359 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2360 sleepTimeShift++; 2361 } 2362 } else { 2363 sleepTime = idleSleepTime; 2364 } 2365 } else if (mBytesWritten != 0 || 2366 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2367 memset (mMixBuffer, 0, mixBufferSize); 2368 sleepTime = 0; 2369 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2370 } 2371 // TODO add standby time extension fct of effect tail 2372} 2373 2374// prepareTracks_l() must be called with ThreadBase::mLock held 2375AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2376 Vector< sp<Track> > *tracksToRemove) 2377{ 2378 2379 mixer_state mixerStatus = MIXER_IDLE; 2380 // find out which tracks need to be processed 2381 size_t count = mActiveTracks.size(); 2382 size_t mixedTracks = 0; 2383 size_t tracksWithEffect = 0; 2384 2385 float masterVolume = mMasterVolume; 2386 bool masterMute = mMasterMute; 2387 2388 if (masterMute) { 2389 masterVolume = 0; 2390 } 2391 // Delegate master volume control to effect in output mix effect chain if needed 2392 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2393 if (chain != 0) { 2394 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2395 chain->setVolume_l(&v, &v); 2396 masterVolume = (float)((v + (1 << 23)) >> 24); 2397 chain.clear(); 2398 } 2399 2400 for (size_t i=0 ; i<count ; i++) { 2401 sp<Track> t = mActiveTracks[i].promote(); 2402 if (t == 0) continue; 2403 2404 // this const just means the local variable doesn't change 2405 Track* const track = t.get(); 2406 audio_track_cblk_t* cblk = track->cblk(); 2407 2408 // The first time a track is added we wait 2409 // for all its buffers to be filled before processing it 2410 int name = track->name(); 2411 // make sure that we have enough frames to mix one full buffer. 2412 // enforce this condition only once to enable draining the buffer in case the client 2413 // app does not call stop() and relies on underrun to stop: 2414 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2415 // during last round 2416 uint32_t minFrames = 1; 2417 if (!track->isStopped() && !track->isPausing() && 2418 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2419 if (t->sampleRate() == (int)mSampleRate) { 2420 minFrames = mFrameCount; 2421 } else { 2422 // +1 for rounding and +1 for additional sample needed for interpolation 2423 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2424 // add frames already consumed but not yet released by the resampler 2425 // because cblk->framesReady() will include these frames 2426 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2427 // the minimum track buffer size is normally twice the number of frames necessary 2428 // to fill one buffer and the resampler should not leave more than one buffer worth 2429 // of unreleased frames after each pass, but just in case... 2430 ALOG_ASSERT(minFrames <= cblk->frameCount); 2431 } 2432 } 2433 if ((track->framesReady() >= minFrames) && track->isReady() && 2434 !track->isPaused() && !track->isTerminated()) 2435 { 2436 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2437 2438 mixedTracks++; 2439 2440 // track->mainBuffer() != mMixBuffer means there is an effect chain 2441 // connected to the track 2442 chain.clear(); 2443 if (track->mainBuffer() != mMixBuffer) { 2444 chain = getEffectChain_l(track->sessionId()); 2445 // Delegate volume control to effect in track effect chain if needed 2446 if (chain != 0) { 2447 tracksWithEffect++; 2448 } else { 2449 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2450 name, track->sessionId()); 2451 } 2452 } 2453 2454 2455 int param = AudioMixer::VOLUME; 2456 if (track->mFillingUpStatus == Track::FS_FILLED) { 2457 // no ramp for the first volume setting 2458 track->mFillingUpStatus = Track::FS_ACTIVE; 2459 if (track->mState == TrackBase::RESUMING) { 2460 track->mState = TrackBase::ACTIVE; 2461 param = AudioMixer::RAMP_VOLUME; 2462 } 2463 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2464 } else if (cblk->server != 0) { 2465 // If the track is stopped before the first frame was mixed, 2466 // do not apply ramp 2467 param = AudioMixer::RAMP_VOLUME; 2468 } 2469 2470 // compute volume for this track 2471 uint32_t vl, vr, va; 2472 if (track->isMuted() || track->isPausing() || 2473 mStreamTypes[track->streamType()].mute) { 2474 vl = vr = va = 0; 2475 if (track->isPausing()) { 2476 track->setPaused(); 2477 } 2478 } else { 2479 2480 // read original volumes with volume control 2481 float typeVolume = mStreamTypes[track->streamType()].volume; 2482 float v = masterVolume * typeVolume; 2483 uint32_t vlr = cblk->getVolumeLR(); 2484 vl = vlr & 0xFFFF; 2485 vr = vlr >> 16; 2486 // track volumes come from shared memory, so can't be trusted and must be clamped 2487 if (vl > MAX_GAIN_INT) { 2488 ALOGV("Track left volume out of range: %04X", vl); 2489 vl = MAX_GAIN_INT; 2490 } 2491 if (vr > MAX_GAIN_INT) { 2492 ALOGV("Track right volume out of range: %04X", vr); 2493 vr = MAX_GAIN_INT; 2494 } 2495 // now apply the master volume and stream type volume 2496 vl = (uint32_t)(v * vl) << 12; 2497 vr = (uint32_t)(v * vr) << 12; 2498 // assuming master volume and stream type volume each go up to 1.0, 2499 // vl and vr are now in 8.24 format 2500 2501 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2502 // send level comes from shared memory and so may be corrupt 2503 if (sendLevel > MAX_GAIN_INT) { 2504 ALOGV("Track send level out of range: %04X", sendLevel); 2505 sendLevel = MAX_GAIN_INT; 2506 } 2507 va = (uint32_t)(v * sendLevel); 2508 } 2509 // Delegate volume control to effect in track effect chain if needed 2510 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2511 // Do not ramp volume if volume is controlled by effect 2512 param = AudioMixer::VOLUME; 2513 track->mHasVolumeController = true; 2514 } else { 2515 // force no volume ramp when volume controller was just disabled or removed 2516 // from effect chain to avoid volume spike 2517 if (track->mHasVolumeController) { 2518 param = AudioMixer::VOLUME; 2519 } 2520 track->mHasVolumeController = false; 2521 } 2522 2523 // Convert volumes from 8.24 to 4.12 format 2524 // This additional clamping is needed in case chain->setVolume_l() overshot 2525 vl = (vl + (1 << 11)) >> 12; 2526 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2527 vr = (vr + (1 << 11)) >> 12; 2528 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2529 2530 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2531 2532 // XXX: these things DON'T need to be done each time 2533 mAudioMixer->setBufferProvider(name, track); 2534 mAudioMixer->enable(name); 2535 2536 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2537 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2538 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2539 mAudioMixer->setParameter( 2540 name, 2541 AudioMixer::TRACK, 2542 AudioMixer::FORMAT, (void *)track->format()); 2543 mAudioMixer->setParameter( 2544 name, 2545 AudioMixer::TRACK, 2546 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2547 mAudioMixer->setParameter( 2548 name, 2549 AudioMixer::RESAMPLE, 2550 AudioMixer::SAMPLE_RATE, 2551 (void *)(cblk->sampleRate)); 2552 mAudioMixer->setParameter( 2553 name, 2554 AudioMixer::TRACK, 2555 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2556 mAudioMixer->setParameter( 2557 name, 2558 AudioMixer::TRACK, 2559 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2560 2561 // reset retry count 2562 track->mRetryCount = kMaxTrackRetries; 2563 2564 // If one track is ready, set the mixer ready if: 2565 // - the mixer was not ready during previous round OR 2566 // - no other track is not ready 2567 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2568 mixerStatus != MIXER_TRACKS_ENABLED) { 2569 mixerStatus = MIXER_TRACKS_READY; 2570 } 2571 } else { 2572 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2573 if (track->isStopped()) { 2574 track->reset(); 2575 } 2576 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2577 // We have consumed all the buffers of this track. 2578 // Remove it from the list of active tracks. 2579 // TODO: use actual buffer filling status instead of latency when available from 2580 // audio HAL 2581 size_t audioHALFrames = 2582 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2583 size_t framesWritten = 2584 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2585 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2586 tracksToRemove->add(track); 2587 } 2588 } else { 2589 // No buffers for this track. Give it a few chances to 2590 // fill a buffer, then remove it from active list. 2591 if (--(track->mRetryCount) <= 0) { 2592 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2593 tracksToRemove->add(track); 2594 // indicate to client process that the track was disabled because of underrun 2595 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2596 // If one track is not ready, mark the mixer also not ready if: 2597 // - the mixer was ready during previous round OR 2598 // - no other track is ready 2599 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2600 mixerStatus != MIXER_TRACKS_READY) { 2601 mixerStatus = MIXER_TRACKS_ENABLED; 2602 } 2603 } 2604 mAudioMixer->disable(name); 2605 } 2606 } 2607 2608 // remove all the tracks that need to be... 2609 count = tracksToRemove->size(); 2610 if (CC_UNLIKELY(count)) { 2611 for (size_t i=0 ; i<count ; i++) { 2612 const sp<Track>& track = tracksToRemove->itemAt(i); 2613 mActiveTracks.remove(track); 2614 if (track->mainBuffer() != mMixBuffer) { 2615 chain = getEffectChain_l(track->sessionId()); 2616 if (chain != 0) { 2617 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2618 chain->decActiveTrackCnt(); 2619 } 2620 } 2621 if (track->isTerminated()) { 2622 removeTrack_l(track); 2623 } 2624 } 2625 } 2626 2627 // mix buffer must be cleared if all tracks are connected to an 2628 // effect chain as in this case the mixer will not write to 2629 // mix buffer and track effects will accumulate into it 2630 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2631 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2632 } 2633 2634 return mixerStatus; 2635} 2636 2637/* 2638The derived values that are cached: 2639 - mixBufferSize from frame count * frame size 2640 - activeSleepTime from activeSleepTimeUs() 2641 - idleSleepTime from idleSleepTimeUs() 2642 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2643 - maxPeriod from frame count and sample rate (MIXER only) 2644 2645The parameters that affect these derived values are: 2646 - frame count 2647 - frame size 2648 - sample rate 2649 - device type: A2DP or not 2650 - device latency 2651 - format: PCM or not 2652 - active sleep time 2653 - idle sleep time 2654*/ 2655 2656void AudioFlinger::PlaybackThread::cacheParameters_l() 2657{ 2658 mixBufferSize = mFrameCount * mFrameSize; 2659 activeSleepTime = activeSleepTimeUs(); 2660 idleSleepTime = idleSleepTimeUs(); 2661} 2662 2663void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2664{ 2665 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2666 this, streamType, mTracks.size()); 2667 Mutex::Autolock _l(mLock); 2668 2669 size_t size = mTracks.size(); 2670 for (size_t i = 0; i < size; i++) { 2671 sp<Track> t = mTracks[i]; 2672 if (t->streamType() == streamType) { 2673 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2674 t->mCblk->cv.signal(); 2675 } 2676 } 2677} 2678 2679// getTrackName_l() must be called with ThreadBase::mLock held 2680int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 2681{ 2682 return mAudioMixer->getTrackName(channelMask); 2683} 2684 2685// deleteTrackName_l() must be called with ThreadBase::mLock held 2686void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2687{ 2688 ALOGV("remove track (%d) and delete from mixer", name); 2689 mAudioMixer->deleteTrackName(name); 2690} 2691 2692// checkForNewParameters_l() must be called with ThreadBase::mLock held 2693bool AudioFlinger::MixerThread::checkForNewParameters_l() 2694{ 2695 bool reconfig = false; 2696 2697 while (!mNewParameters.isEmpty()) { 2698 status_t status = NO_ERROR; 2699 String8 keyValuePair = mNewParameters[0]; 2700 AudioParameter param = AudioParameter(keyValuePair); 2701 int value; 2702 2703 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2704 reconfig = true; 2705 } 2706 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2707 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2708 status = BAD_VALUE; 2709 } else { 2710 reconfig = true; 2711 } 2712 } 2713 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2714 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2715 status = BAD_VALUE; 2716 } else { 2717 reconfig = true; 2718 } 2719 } 2720 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2721 // do not accept frame count changes if tracks are open as the track buffer 2722 // size depends on frame count and correct behavior would not be guaranteed 2723 // if frame count is changed after track creation 2724 if (!mTracks.isEmpty()) { 2725 status = INVALID_OPERATION; 2726 } else { 2727 reconfig = true; 2728 } 2729 } 2730 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2731#ifdef ADD_BATTERY_DATA 2732 // when changing the audio output device, call addBatteryData to notify 2733 // the change 2734 if ((int)mDevice != value) { 2735 uint32_t params = 0; 2736 // check whether speaker is on 2737 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2738 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2739 } 2740 2741 int deviceWithoutSpeaker 2742 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2743 // check if any other device (except speaker) is on 2744 if (value & deviceWithoutSpeaker ) { 2745 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2746 } 2747 2748 if (params != 0) { 2749 addBatteryData(params); 2750 } 2751 } 2752#endif 2753 2754 // forward device change to effects that have requested to be 2755 // aware of attached audio device. 2756 mDevice = (uint32_t)value; 2757 for (size_t i = 0; i < mEffectChains.size(); i++) { 2758 mEffectChains[i]->setDevice_l(mDevice); 2759 } 2760 } 2761 2762 if (status == NO_ERROR) { 2763 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2764 keyValuePair.string()); 2765 if (!mStandby && status == INVALID_OPERATION) { 2766 mOutput->stream->common.standby(&mOutput->stream->common); 2767 mStandby = true; 2768 mBytesWritten = 0; 2769 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2770 keyValuePair.string()); 2771 } 2772 if (status == NO_ERROR && reconfig) { 2773 delete mAudioMixer; 2774 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2775 mAudioMixer = NULL; 2776 readOutputParameters(); 2777 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2778 for (size_t i = 0; i < mTracks.size() ; i++) { 2779 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 2780 if (name < 0) break; 2781 mTracks[i]->mName = name; 2782 // limit track sample rate to 2 x new output sample rate 2783 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2784 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2785 } 2786 } 2787 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2788 } 2789 } 2790 2791 mNewParameters.removeAt(0); 2792 2793 mParamStatus = status; 2794 mParamCond.signal(); 2795 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2796 // already timed out waiting for the status and will never signal the condition. 2797 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2798 } 2799 return reconfig; 2800} 2801 2802status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2803{ 2804 const size_t SIZE = 256; 2805 char buffer[SIZE]; 2806 String8 result; 2807 2808 PlaybackThread::dumpInternals(fd, args); 2809 2810 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2811 result.append(buffer); 2812 write(fd, result.string(), result.size()); 2813 return NO_ERROR; 2814} 2815 2816uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 2817{ 2818 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2819} 2820 2821uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 2822{ 2823 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2824} 2825 2826void AudioFlinger::MixerThread::cacheParameters_l() 2827{ 2828 PlaybackThread::cacheParameters_l(); 2829 2830 // FIXME: Relaxed timing because of a certain device that can't meet latency 2831 // Should be reduced to 2x after the vendor fixes the driver issue 2832 // increase threshold again due to low power audio mode. The way this warning 2833 // threshold is calculated and its usefulness should be reconsidered anyway. 2834 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2835} 2836 2837// ---------------------------------------------------------------------------- 2838AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2839 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2840 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2841 // mLeftVolFloat, mRightVolFloat 2842 // mLeftVolShort, mRightVolShort 2843{ 2844} 2845 2846AudioFlinger::DirectOutputThread::~DirectOutputThread() 2847{ 2848} 2849 2850AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2851 Vector< sp<Track> > *tracksToRemove 2852) 2853{ 2854 sp<Track> trackToRemove; 2855 2856 mixer_state mixerStatus = MIXER_IDLE; 2857 2858 // find out which tracks need to be processed 2859 if (mActiveTracks.size() != 0) { 2860 sp<Track> t = mActiveTracks[0].promote(); 2861 // The track died recently 2862 if (t == 0) return MIXER_IDLE; 2863 2864 Track* const track = t.get(); 2865 audio_track_cblk_t* cblk = track->cblk(); 2866 2867 // The first time a track is added we wait 2868 // for all its buffers to be filled before processing it 2869 if (cblk->framesReady() && track->isReady() && 2870 !track->isPaused() && !track->isTerminated()) 2871 { 2872 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2873 2874 if (track->mFillingUpStatus == Track::FS_FILLED) { 2875 track->mFillingUpStatus = Track::FS_ACTIVE; 2876 mLeftVolFloat = mRightVolFloat = 0; 2877 mLeftVolShort = mRightVolShort = 0; 2878 if (track->mState == TrackBase::RESUMING) { 2879 track->mState = TrackBase::ACTIVE; 2880 rampVolume = true; 2881 } 2882 } else if (cblk->server != 0) { 2883 // If the track is stopped before the first frame was mixed, 2884 // do not apply ramp 2885 rampVolume = true; 2886 } 2887 // compute volume for this track 2888 float left, right; 2889 if (track->isMuted() || mMasterMute || track->isPausing() || 2890 mStreamTypes[track->streamType()].mute) { 2891 left = right = 0; 2892 if (track->isPausing()) { 2893 track->setPaused(); 2894 } 2895 } else { 2896 float typeVolume = mStreamTypes[track->streamType()].volume; 2897 float v = mMasterVolume * typeVolume; 2898 uint32_t vlr = cblk->getVolumeLR(); 2899 float v_clamped = v * (vlr & 0xFFFF); 2900 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2901 left = v_clamped/MAX_GAIN; 2902 v_clamped = v * (vlr >> 16); 2903 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2904 right = v_clamped/MAX_GAIN; 2905 } 2906 2907 if (left != mLeftVolFloat || right != mRightVolFloat) { 2908 mLeftVolFloat = left; 2909 mRightVolFloat = right; 2910 2911 // If audio HAL implements volume control, 2912 // force software volume to nominal value 2913 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2914 left = 1.0f; 2915 right = 1.0f; 2916 } 2917 2918 // Convert volumes from float to 8.24 2919 uint32_t vl = (uint32_t)(left * (1 << 24)); 2920 uint32_t vr = (uint32_t)(right * (1 << 24)); 2921 2922 // Delegate volume control to effect in track effect chain if needed 2923 // only one effect chain can be present on DirectOutputThread, so if 2924 // there is one, the track is connected to it 2925 if (!mEffectChains.isEmpty()) { 2926 // Do not ramp volume if volume is controlled by effect 2927 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2928 rampVolume = false; 2929 } 2930 } 2931 2932 // Convert volumes from 8.24 to 4.12 format 2933 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2934 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2935 leftVol = (uint16_t)v_clamped; 2936 v_clamped = (vr + (1 << 11)) >> 12; 2937 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2938 rightVol = (uint16_t)v_clamped; 2939 } else { 2940 leftVol = mLeftVolShort; 2941 rightVol = mRightVolShort; 2942 rampVolume = false; 2943 } 2944 2945 // reset retry count 2946 track->mRetryCount = kMaxTrackRetriesDirect; 2947 mActiveTrack = t; 2948 mixerStatus = MIXER_TRACKS_READY; 2949 } else { 2950 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2951 if (track->isStopped()) { 2952 track->reset(); 2953 } 2954 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2955 // We have consumed all the buffers of this track. 2956 // Remove it from the list of active tracks. 2957 // TODO: implement behavior for compressed audio 2958 size_t audioHALFrames = 2959 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2960 size_t framesWritten = 2961 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2962 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2963 trackToRemove = track; 2964 } 2965 } else { 2966 // No buffers for this track. Give it a few chances to 2967 // fill a buffer, then remove it from active list. 2968 if (--(track->mRetryCount) <= 0) { 2969 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2970 trackToRemove = track; 2971 } else { 2972 mixerStatus = MIXER_TRACKS_ENABLED; 2973 } 2974 } 2975 } 2976 } 2977 2978 // FIXME merge this with similar code for removing multiple tracks 2979 // remove all the tracks that need to be... 2980 if (CC_UNLIKELY(trackToRemove != 0)) { 2981 tracksToRemove->add(trackToRemove); 2982 mActiveTracks.remove(trackToRemove); 2983 if (!mEffectChains.isEmpty()) { 2984 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2985 trackToRemove->sessionId()); 2986 mEffectChains[0]->decActiveTrackCnt(); 2987 } 2988 if (trackToRemove->isTerminated()) { 2989 removeTrack_l(trackToRemove); 2990 } 2991 } 2992 2993 return mixerStatus; 2994} 2995 2996void AudioFlinger::DirectOutputThread::threadLoop_mix() 2997{ 2998 AudioBufferProvider::Buffer buffer; 2999 size_t frameCount = mFrameCount; 3000 int8_t *curBuf = (int8_t *)mMixBuffer; 3001 // output audio to hardware 3002 while (frameCount) { 3003 buffer.frameCount = frameCount; 3004 mActiveTrack->getNextBuffer(&buffer); 3005 if (CC_UNLIKELY(buffer.raw == NULL)) { 3006 memset(curBuf, 0, frameCount * mFrameSize); 3007 break; 3008 } 3009 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3010 frameCount -= buffer.frameCount; 3011 curBuf += buffer.frameCount * mFrameSize; 3012 mActiveTrack->releaseBuffer(&buffer); 3013 } 3014 sleepTime = 0; 3015 standbyTime = systemTime() + standbyDelay; 3016 mActiveTrack.clear(); 3017 3018 // apply volume 3019 3020 // Do not apply volume on compressed audio 3021 if (!audio_is_linear_pcm(mFormat)) { 3022 return; 3023 } 3024 3025 // convert to signed 16 bit before volume calculation 3026 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3027 size_t count = mFrameCount * mChannelCount; 3028 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3029 int16_t *dst = mMixBuffer + count-1; 3030 while (count--) { 3031 *dst-- = (int16_t)(*src--^0x80) << 8; 3032 } 3033 } 3034 3035 frameCount = mFrameCount; 3036 int16_t *out = mMixBuffer; 3037 if (rampVolume) { 3038 if (mChannelCount == 1) { 3039 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3040 int32_t vlInc = d / (int32_t)frameCount; 3041 int32_t vl = ((int32_t)mLeftVolShort << 16); 3042 do { 3043 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3044 out++; 3045 vl += vlInc; 3046 } while (--frameCount); 3047 3048 } else { 3049 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3050 int32_t vlInc = d / (int32_t)frameCount; 3051 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3052 int32_t vrInc = d / (int32_t)frameCount; 3053 int32_t vl = ((int32_t)mLeftVolShort << 16); 3054 int32_t vr = ((int32_t)mRightVolShort << 16); 3055 do { 3056 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3057 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3058 out += 2; 3059 vl += vlInc; 3060 vr += vrInc; 3061 } while (--frameCount); 3062 } 3063 } else { 3064 if (mChannelCount == 1) { 3065 do { 3066 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3067 out++; 3068 } while (--frameCount); 3069 } else { 3070 do { 3071 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3072 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3073 out += 2; 3074 } while (--frameCount); 3075 } 3076 } 3077 3078 // convert back to unsigned 8 bit after volume calculation 3079 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3080 size_t count = mFrameCount * mChannelCount; 3081 int16_t *src = mMixBuffer; 3082 uint8_t *dst = (uint8_t *)mMixBuffer; 3083 while (count--) { 3084 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3085 } 3086 } 3087 3088 mLeftVolShort = leftVol; 3089 mRightVolShort = rightVol; 3090} 3091 3092void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3093{ 3094 if (sleepTime == 0) { 3095 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3096 sleepTime = activeSleepTime; 3097 } else { 3098 sleepTime = idleSleepTime; 3099 } 3100 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3101 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3102 sleepTime = 0; 3103 } 3104} 3105 3106// getTrackName_l() must be called with ThreadBase::mLock held 3107int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3108{ 3109 return 0; 3110} 3111 3112// deleteTrackName_l() must be called with ThreadBase::mLock held 3113void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3114{ 3115} 3116 3117// checkForNewParameters_l() must be called with ThreadBase::mLock held 3118bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3119{ 3120 bool reconfig = false; 3121 3122 while (!mNewParameters.isEmpty()) { 3123 status_t status = NO_ERROR; 3124 String8 keyValuePair = mNewParameters[0]; 3125 AudioParameter param = AudioParameter(keyValuePair); 3126 int value; 3127 3128 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3129 // do not accept frame count changes if tracks are open as the track buffer 3130 // size depends on frame count and correct behavior would not be garantied 3131 // if frame count is changed after track creation 3132 if (!mTracks.isEmpty()) { 3133 status = INVALID_OPERATION; 3134 } else { 3135 reconfig = true; 3136 } 3137 } 3138 if (status == NO_ERROR) { 3139 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3140 keyValuePair.string()); 3141 if (!mStandby && status == INVALID_OPERATION) { 3142 mOutput->stream->common.standby(&mOutput->stream->common); 3143 mStandby = true; 3144 mBytesWritten = 0; 3145 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3146 keyValuePair.string()); 3147 } 3148 if (status == NO_ERROR && reconfig) { 3149 readOutputParameters(); 3150 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3151 } 3152 } 3153 3154 mNewParameters.removeAt(0); 3155 3156 mParamStatus = status; 3157 mParamCond.signal(); 3158 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3159 // already timed out waiting for the status and will never signal the condition. 3160 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3161 } 3162 return reconfig; 3163} 3164 3165uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3166{ 3167 uint32_t time; 3168 if (audio_is_linear_pcm(mFormat)) { 3169 time = PlaybackThread::activeSleepTimeUs(); 3170 } else { 3171 time = 10000; 3172 } 3173 return time; 3174} 3175 3176uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3177{ 3178 uint32_t time; 3179 if (audio_is_linear_pcm(mFormat)) { 3180 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3181 } else { 3182 time = 10000; 3183 } 3184 return time; 3185} 3186 3187uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3188{ 3189 uint32_t time; 3190 if (audio_is_linear_pcm(mFormat)) { 3191 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3192 } else { 3193 time = 10000; 3194 } 3195 return time; 3196} 3197 3198void AudioFlinger::DirectOutputThread::cacheParameters_l() 3199{ 3200 PlaybackThread::cacheParameters_l(); 3201 3202 // use shorter standby delay as on normal output to release 3203 // hardware resources as soon as possible 3204 standbyDelay = microseconds(activeSleepTime*2); 3205} 3206 3207// ---------------------------------------------------------------------------- 3208 3209AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3210 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3211 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3212 mWaitTimeMs(UINT_MAX) 3213{ 3214 addOutputTrack(mainThread); 3215} 3216 3217AudioFlinger::DuplicatingThread::~DuplicatingThread() 3218{ 3219 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3220 mOutputTracks[i]->destroy(); 3221 } 3222} 3223 3224void AudioFlinger::DuplicatingThread::threadLoop_mix() 3225{ 3226 // mix buffers... 3227 if (outputsReady(outputTracks)) { 3228 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3229 } else { 3230 memset(mMixBuffer, 0, mixBufferSize); 3231 } 3232 sleepTime = 0; 3233 writeFrames = mFrameCount; 3234} 3235 3236void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3237{ 3238 if (sleepTime == 0) { 3239 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3240 sleepTime = activeSleepTime; 3241 } else { 3242 sleepTime = idleSleepTime; 3243 } 3244 } else if (mBytesWritten != 0) { 3245 // flush remaining overflow buffers in output tracks 3246 for (size_t i = 0; i < outputTracks.size(); i++) { 3247 if (outputTracks[i]->isActive()) { 3248 sleepTime = 0; 3249 writeFrames = 0; 3250 memset(mMixBuffer, 0, mixBufferSize); 3251 break; 3252 } 3253 } 3254 } 3255} 3256 3257void AudioFlinger::DuplicatingThread::threadLoop_write() 3258{ 3259 standbyTime = systemTime() + standbyDelay; 3260 for (size_t i = 0; i < outputTracks.size(); i++) { 3261 outputTracks[i]->write(mMixBuffer, writeFrames); 3262 } 3263 mBytesWritten += mixBufferSize; 3264} 3265 3266void AudioFlinger::DuplicatingThread::threadLoop_standby() 3267{ 3268 // DuplicatingThread implements standby by stopping all tracks 3269 for (size_t i = 0; i < outputTracks.size(); i++) { 3270 outputTracks[i]->stop(); 3271 } 3272} 3273 3274void AudioFlinger::DuplicatingThread::saveOutputTracks() 3275{ 3276 outputTracks = mOutputTracks; 3277} 3278 3279void AudioFlinger::DuplicatingThread::clearOutputTracks() 3280{ 3281 outputTracks.clear(); 3282} 3283 3284void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3285{ 3286 Mutex::Autolock _l(mLock); 3287 // FIXME explain this formula 3288 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3289 OutputTrack *outputTrack = new OutputTrack(thread, 3290 this, 3291 mSampleRate, 3292 mFormat, 3293 mChannelMask, 3294 frameCount); 3295 if (outputTrack->cblk() != NULL) { 3296 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3297 mOutputTracks.add(outputTrack); 3298 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3299 updateWaitTime_l(); 3300 } 3301} 3302 3303void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3304{ 3305 Mutex::Autolock _l(mLock); 3306 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3307 if (mOutputTracks[i]->thread() == thread) { 3308 mOutputTracks[i]->destroy(); 3309 mOutputTracks.removeAt(i); 3310 updateWaitTime_l(); 3311 return; 3312 } 3313 } 3314 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3315} 3316 3317// caller must hold mLock 3318void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3319{ 3320 mWaitTimeMs = UINT_MAX; 3321 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3322 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3323 if (strong != 0) { 3324 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3325 if (waitTimeMs < mWaitTimeMs) { 3326 mWaitTimeMs = waitTimeMs; 3327 } 3328 } 3329 } 3330} 3331 3332 3333bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3334{ 3335 for (size_t i = 0; i < outputTracks.size(); i++) { 3336 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3337 if (thread == 0) { 3338 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3339 return false; 3340 } 3341 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3342 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3343 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3344 return false; 3345 } 3346 } 3347 return true; 3348} 3349 3350uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3351{ 3352 return (mWaitTimeMs * 1000) / 2; 3353} 3354 3355void AudioFlinger::DuplicatingThread::cacheParameters_l() 3356{ 3357 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3358 updateWaitTime_l(); 3359 3360 MixerThread::cacheParameters_l(); 3361} 3362 3363// ---------------------------------------------------------------------------- 3364 3365// TrackBase constructor must be called with AudioFlinger::mLock held 3366AudioFlinger::ThreadBase::TrackBase::TrackBase( 3367 ThreadBase *thread, 3368 const sp<Client>& client, 3369 uint32_t sampleRate, 3370 audio_format_t format, 3371 uint32_t channelMask, 3372 int frameCount, 3373 const sp<IMemory>& sharedBuffer, 3374 int sessionId) 3375 : RefBase(), 3376 mThread(thread), 3377 mClient(client), 3378 mCblk(NULL), 3379 // mBuffer 3380 // mBufferEnd 3381 mFrameCount(0), 3382 mState(IDLE), 3383 mFormat(format), 3384 mStepServerFailed(false), 3385 mSessionId(sessionId) 3386 // mChannelCount 3387 // mChannelMask 3388{ 3389 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3390 3391 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3392 size_t size = sizeof(audio_track_cblk_t); 3393 uint8_t channelCount = popcount(channelMask); 3394 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3395 if (sharedBuffer == 0) { 3396 size += bufferSize; 3397 } 3398 3399 if (client != NULL) { 3400 mCblkMemory = client->heap()->allocate(size); 3401 if (mCblkMemory != 0) { 3402 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3403 if (mCblk != NULL) { // construct the shared structure in-place. 3404 new(mCblk) audio_track_cblk_t(); 3405 // clear all buffers 3406 mCblk->frameCount = frameCount; 3407 mCblk->sampleRate = sampleRate; 3408// uncomment the following lines to quickly test 32-bit wraparound 3409// mCblk->user = 0xffff0000; 3410// mCblk->server = 0xffff0000; 3411// mCblk->userBase = 0xffff0000; 3412// mCblk->serverBase = 0xffff0000; 3413 mChannelCount = channelCount; 3414 mChannelMask = channelMask; 3415 if (sharedBuffer == 0) { 3416 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3417 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3418 // Force underrun condition to avoid false underrun callback until first data is 3419 // written to buffer (other flags are cleared) 3420 mCblk->flags = CBLK_UNDERRUN_ON; 3421 } else { 3422 mBuffer = sharedBuffer->pointer(); 3423 } 3424 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3425 } 3426 } else { 3427 ALOGE("not enough memory for AudioTrack size=%u", size); 3428 client->heap()->dump("AudioTrack"); 3429 return; 3430 } 3431 } else { 3432 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3433 // construct the shared structure in-place. 3434 new(mCblk) audio_track_cblk_t(); 3435 // clear all buffers 3436 mCblk->frameCount = frameCount; 3437 mCblk->sampleRate = sampleRate; 3438// uncomment the following lines to quickly test 32-bit wraparound 3439// mCblk->user = 0xffff0000; 3440// mCblk->server = 0xffff0000; 3441// mCblk->userBase = 0xffff0000; 3442// mCblk->serverBase = 0xffff0000; 3443 mChannelCount = channelCount; 3444 mChannelMask = channelMask; 3445 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3446 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3447 // Force underrun condition to avoid false underrun callback until first data is 3448 // written to buffer (other flags are cleared) 3449 mCblk->flags = CBLK_UNDERRUN_ON; 3450 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3451 } 3452} 3453 3454AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3455{ 3456 if (mCblk != NULL) { 3457 if (mClient == 0) { 3458 delete mCblk; 3459 } else { 3460 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3461 } 3462 } 3463 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3464 if (mClient != 0) { 3465 // Client destructor must run with AudioFlinger mutex locked 3466 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3467 // If the client's reference count drops to zero, the associated destructor 3468 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3469 // relying on the automatic clear() at end of scope. 3470 mClient.clear(); 3471 } 3472} 3473 3474// AudioBufferProvider interface 3475// getNextBuffer() = 0; 3476// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3477void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3478{ 3479 buffer->raw = NULL; 3480 mFrameCount = buffer->frameCount; 3481 (void) step(); // ignore return value of step() 3482 buffer->frameCount = 0; 3483} 3484 3485bool AudioFlinger::ThreadBase::TrackBase::step() { 3486 bool result; 3487 audio_track_cblk_t* cblk = this->cblk(); 3488 3489 result = cblk->stepServer(mFrameCount); 3490 if (!result) { 3491 ALOGV("stepServer failed acquiring cblk mutex"); 3492 mStepServerFailed = true; 3493 } 3494 return result; 3495} 3496 3497void AudioFlinger::ThreadBase::TrackBase::reset() { 3498 audio_track_cblk_t* cblk = this->cblk(); 3499 3500 cblk->user = 0; 3501 cblk->server = 0; 3502 cblk->userBase = 0; 3503 cblk->serverBase = 0; 3504 mStepServerFailed = false; 3505 ALOGV("TrackBase::reset"); 3506} 3507 3508int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3509 return (int)mCblk->sampleRate; 3510} 3511 3512void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3513 audio_track_cblk_t* cblk = this->cblk(); 3514 size_t frameSize = cblk->frameSize; 3515 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3516 int8_t *bufferEnd = bufferStart + frames * frameSize; 3517 3518 // Check validity of returned pointer in case the track control block would have been corrupted. 3519 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 3520 "TrackBase::getBuffer buffer out of range:\n" 3521 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 3522 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 3523 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3524 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 3525 3526 return bufferStart; 3527} 3528 3529status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3530{ 3531 mSyncEvents.add(event); 3532 return NO_ERROR; 3533} 3534 3535// ---------------------------------------------------------------------------- 3536 3537// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3538AudioFlinger::PlaybackThread::Track::Track( 3539 PlaybackThread *thread, 3540 const sp<Client>& client, 3541 audio_stream_type_t streamType, 3542 uint32_t sampleRate, 3543 audio_format_t format, 3544 uint32_t channelMask, 3545 int frameCount, 3546 const sp<IMemory>& sharedBuffer, 3547 int sessionId, 3548 IAudioFlinger::track_flags_t flags) 3549 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3550 mMute(false), 3551 // mFillingUpStatus ? 3552 // mRetryCount initialized later when needed 3553 mSharedBuffer(sharedBuffer), 3554 mStreamType(streamType), 3555 mName(-1), // see note below 3556 mMainBuffer(thread->mixBuffer()), 3557 mAuxBuffer(NULL), 3558 mAuxEffectId(0), mHasVolumeController(false), 3559 mPresentationCompleteFrames(0), 3560 mFlags(flags) 3561{ 3562 if (mCblk != NULL) { 3563 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3564 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3565 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3566 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3567 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 3568 if (mName < 0) { 3569 ALOGE("no more track names available"); 3570 } 3571 } 3572 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3573} 3574 3575AudioFlinger::PlaybackThread::Track::~Track() 3576{ 3577 ALOGV("PlaybackThread::Track destructor"); 3578 sp<ThreadBase> thread = mThread.promote(); 3579 if (thread != 0) { 3580 Mutex::Autolock _l(thread->mLock); 3581 mState = TERMINATED; 3582 } 3583} 3584 3585void AudioFlinger::PlaybackThread::Track::destroy() 3586{ 3587 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3588 // by removing it from mTracks vector, so there is a risk that this Tracks's 3589 // destructor is called. As the destructor needs to lock mLock, 3590 // we must acquire a strong reference on this Track before locking mLock 3591 // here so that the destructor is called only when exiting this function. 3592 // On the other hand, as long as Track::destroy() is only called by 3593 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3594 // this Track with its member mTrack. 3595 sp<Track> keep(this); 3596 { // scope for mLock 3597 sp<ThreadBase> thread = mThread.promote(); 3598 if (thread != 0) { 3599 if (!isOutputTrack()) { 3600 if (mState == ACTIVE || mState == RESUMING) { 3601 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3602 3603#ifdef ADD_BATTERY_DATA 3604 // to track the speaker usage 3605 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3606#endif 3607 } 3608 AudioSystem::releaseOutput(thread->id()); 3609 } 3610 Mutex::Autolock _l(thread->mLock); 3611 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3612 playbackThread->destroyTrack_l(this); 3613 } 3614 } 3615} 3616 3617void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3618{ 3619 uint32_t vlr = mCblk->getVolumeLR(); 3620 snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", 3621 mName - AudioMixer::TRACK0, 3622 (mClient == 0) ? getpid_cached : mClient->pid(), 3623 mStreamType, 3624 mFormat, 3625 mChannelMask, 3626 mSessionId, 3627 mFrameCount, 3628 mState, 3629 mMute, 3630 mFillingUpStatus, 3631 mCblk->sampleRate, 3632 vlr & 0xFFFF, 3633 vlr >> 16, 3634 mCblk->server, 3635 mCblk->user, 3636 (int)mMainBuffer, 3637 (int)mAuxBuffer); 3638} 3639 3640// AudioBufferProvider interface 3641status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3642 AudioBufferProvider::Buffer* buffer, int64_t pts) 3643{ 3644 audio_track_cblk_t* cblk = this->cblk(); 3645 uint32_t framesReady; 3646 uint32_t framesReq = buffer->frameCount; 3647 3648 // Check if last stepServer failed, try to step now 3649 if (mStepServerFailed) { 3650 if (!step()) goto getNextBuffer_exit; 3651 ALOGV("stepServer recovered"); 3652 mStepServerFailed = false; 3653 } 3654 3655 framesReady = cblk->framesReady(); 3656 3657 if (CC_LIKELY(framesReady)) { 3658 uint32_t s = cblk->server; 3659 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3660 3661 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3662 if (framesReq > framesReady) { 3663 framesReq = framesReady; 3664 } 3665 if (framesReq > bufferEnd - s) { 3666 framesReq = bufferEnd - s; 3667 } 3668 3669 buffer->raw = getBuffer(s, framesReq); 3670 if (buffer->raw == NULL) goto getNextBuffer_exit; 3671 3672 buffer->frameCount = framesReq; 3673 return NO_ERROR; 3674 } 3675 3676getNextBuffer_exit: 3677 buffer->raw = NULL; 3678 buffer->frameCount = 0; 3679 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3680 return NOT_ENOUGH_DATA; 3681} 3682 3683uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3684 return mCblk->framesReady(); 3685} 3686 3687bool AudioFlinger::PlaybackThread::Track::isReady() const { 3688 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3689 3690 if (framesReady() >= mCblk->frameCount || 3691 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3692 mFillingUpStatus = FS_FILLED; 3693 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3694 return true; 3695 } 3696 return false; 3697} 3698 3699status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 3700 int triggerSession) 3701{ 3702 status_t status = NO_ERROR; 3703 3704 sp<ThreadBase> thread = mThread.promote(); 3705 if (thread != 0) { 3706 Mutex::Autolock _l(thread->mLock); 3707 track_state state = mState; 3708 // here the track could be either new, or restarted 3709 // in both cases "unstop" the track 3710 if (mState == PAUSED) { 3711 mState = TrackBase::RESUMING; 3712 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3713 } else { 3714 mState = TrackBase::ACTIVE; 3715 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3716 } 3717 3718 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3719 thread->mLock.unlock(); 3720 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3721 thread->mLock.lock(); 3722 3723#ifdef ADD_BATTERY_DATA 3724 // to track the speaker usage 3725 if (status == NO_ERROR) { 3726 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3727 } 3728#endif 3729 } 3730 if (status == NO_ERROR) { 3731 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3732 playbackThread->addTrack_l(this); 3733 } else { 3734 mState = state; 3735 } 3736 } else { 3737 status = BAD_VALUE; 3738 } 3739 return status; 3740} 3741 3742void AudioFlinger::PlaybackThread::Track::stop() 3743{ 3744 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3745 sp<ThreadBase> thread = mThread.promote(); 3746 if (thread != 0) { 3747 Mutex::Autolock _l(thread->mLock); 3748 track_state state = mState; 3749 if (mState > STOPPED) { 3750 mState = STOPPED; 3751 // If the track is not active (PAUSED and buffers full), flush buffers 3752 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3753 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3754 reset(); 3755 } 3756 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3757 } 3758 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3759 thread->mLock.unlock(); 3760 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3761 thread->mLock.lock(); 3762 3763#ifdef ADD_BATTERY_DATA 3764 // to track the speaker usage 3765 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3766#endif 3767 } 3768 } 3769} 3770 3771void AudioFlinger::PlaybackThread::Track::pause() 3772{ 3773 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3774 sp<ThreadBase> thread = mThread.promote(); 3775 if (thread != 0) { 3776 Mutex::Autolock _l(thread->mLock); 3777 if (mState == ACTIVE || mState == RESUMING) { 3778 mState = PAUSING; 3779 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3780 if (!isOutputTrack()) { 3781 thread->mLock.unlock(); 3782 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3783 thread->mLock.lock(); 3784 3785#ifdef ADD_BATTERY_DATA 3786 // to track the speaker usage 3787 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3788#endif 3789 } 3790 } 3791 } 3792} 3793 3794void AudioFlinger::PlaybackThread::Track::flush() 3795{ 3796 ALOGV("flush(%d)", mName); 3797 sp<ThreadBase> thread = mThread.promote(); 3798 if (thread != 0) { 3799 Mutex::Autolock _l(thread->mLock); 3800 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3801 return; 3802 } 3803 // No point remaining in PAUSED state after a flush => go to 3804 // STOPPED state 3805 mState = STOPPED; 3806 3807 // do not reset the track if it is still in the process of being stopped or paused. 3808 // this will be done by prepareTracks_l() when the track is stopped. 3809 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3810 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3811 reset(); 3812 } 3813 } 3814} 3815 3816void AudioFlinger::PlaybackThread::Track::reset() 3817{ 3818 // Do not reset twice to avoid discarding data written just after a flush and before 3819 // the audioflinger thread detects the track is stopped. 3820 if (!mResetDone) { 3821 TrackBase::reset(); 3822 // Force underrun condition to avoid false underrun callback until first data is 3823 // written to buffer 3824 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3825 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3826 mFillingUpStatus = FS_FILLING; 3827 mResetDone = true; 3828 mPresentationCompleteFrames = 0; 3829 } 3830} 3831 3832void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3833{ 3834 mMute = muted; 3835} 3836 3837status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3838{ 3839 status_t status = DEAD_OBJECT; 3840 sp<ThreadBase> thread = mThread.promote(); 3841 if (thread != 0) { 3842 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3843 status = playbackThread->attachAuxEffect(this, EffectId); 3844 } 3845 return status; 3846} 3847 3848void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3849{ 3850 mAuxEffectId = EffectId; 3851 mAuxBuffer = buffer; 3852} 3853 3854bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 3855 size_t audioHalFrames) 3856{ 3857 // a track is considered presented when the total number of frames written to audio HAL 3858 // corresponds to the number of frames written when presentationComplete() is called for the 3859 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 3860 if (mPresentationCompleteFrames == 0) { 3861 mPresentationCompleteFrames = framesWritten + audioHalFrames; 3862 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 3863 mPresentationCompleteFrames, audioHalFrames); 3864 } 3865 if (framesWritten >= mPresentationCompleteFrames) { 3866 ALOGV("presentationComplete() session %d complete: framesWritten %d", 3867 mSessionId, framesWritten); 3868 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 3869 mPresentationCompleteFrames = 0; 3870 return true; 3871 } 3872 return false; 3873} 3874 3875void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 3876{ 3877 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 3878 if (mSyncEvents[i]->type() == type) { 3879 mSyncEvents[i]->trigger(); 3880 mSyncEvents.removeAt(i); 3881 i--; 3882 } 3883 } 3884} 3885 3886 3887// timed audio tracks 3888 3889sp<AudioFlinger::PlaybackThread::TimedTrack> 3890AudioFlinger::PlaybackThread::TimedTrack::create( 3891 PlaybackThread *thread, 3892 const sp<Client>& client, 3893 audio_stream_type_t streamType, 3894 uint32_t sampleRate, 3895 audio_format_t format, 3896 uint32_t channelMask, 3897 int frameCount, 3898 const sp<IMemory>& sharedBuffer, 3899 int sessionId) { 3900 if (!client->reserveTimedTrack()) 3901 return NULL; 3902 3903 return new TimedTrack( 3904 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3905 sharedBuffer, sessionId); 3906} 3907 3908AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3909 PlaybackThread *thread, 3910 const sp<Client>& client, 3911 audio_stream_type_t streamType, 3912 uint32_t sampleRate, 3913 audio_format_t format, 3914 uint32_t channelMask, 3915 int frameCount, 3916 const sp<IMemory>& sharedBuffer, 3917 int sessionId) 3918 : Track(thread, client, streamType, sampleRate, format, channelMask, 3919 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 3920 mQueueHeadInFlight(false), 3921 mTrimQueueHeadOnRelease(false), 3922 mFramesPendingInQueue(0), 3923 mTimedSilenceBuffer(NULL), 3924 mTimedSilenceBufferSize(0), 3925 mTimedAudioOutputOnTime(false), 3926 mMediaTimeTransformValid(false) 3927{ 3928 LocalClock lc; 3929 mLocalTimeFreq = lc.getLocalFreq(); 3930 3931 mLocalTimeToSampleTransform.a_zero = 0; 3932 mLocalTimeToSampleTransform.b_zero = 0; 3933 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3934 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3935 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3936 &mLocalTimeToSampleTransform.a_to_b_denom); 3937 3938 mMediaTimeToSampleTransform.a_zero = 0; 3939 mMediaTimeToSampleTransform.b_zero = 0; 3940 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 3941 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 3942 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 3943 &mMediaTimeToSampleTransform.a_to_b_denom); 3944} 3945 3946AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3947 mClient->releaseTimedTrack(); 3948 delete [] mTimedSilenceBuffer; 3949} 3950 3951status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3952 size_t size, sp<IMemory>* buffer) { 3953 3954 Mutex::Autolock _l(mTimedBufferQueueLock); 3955 3956 trimTimedBufferQueue_l(); 3957 3958 // lazily initialize the shared memory heap for timed buffers 3959 if (mTimedMemoryDealer == NULL) { 3960 const int kTimedBufferHeapSize = 512 << 10; 3961 3962 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3963 "AudioFlingerTimed"); 3964 if (mTimedMemoryDealer == NULL) 3965 return NO_MEMORY; 3966 } 3967 3968 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3969 if (newBuffer == NULL) { 3970 newBuffer = mTimedMemoryDealer->allocate(size); 3971 if (newBuffer == NULL) 3972 return NO_MEMORY; 3973 } 3974 3975 *buffer = newBuffer; 3976 return NO_ERROR; 3977} 3978 3979// caller must hold mTimedBufferQueueLock 3980void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3981 int64_t mediaTimeNow; 3982 { 3983 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3984 if (!mMediaTimeTransformValid) 3985 return; 3986 3987 int64_t targetTimeNow; 3988 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3989 ? mCCHelper.getCommonTime(&targetTimeNow) 3990 : mCCHelper.getLocalTime(&targetTimeNow); 3991 3992 if (OK != res) 3993 return; 3994 3995 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3996 &mediaTimeNow)) { 3997 return; 3998 } 3999 } 4000 4001 size_t trimEnd; 4002 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4003 int64_t bufEnd; 4004 4005 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4006 // We have a next buffer. Just use its PTS as the PTS of the frame 4007 // following the last frame in this buffer. If the stream is sparse 4008 // (ie, there are deliberate gaps left in the stream which should be 4009 // filled with silence by the TimedAudioTrack), then this can result 4010 // in one extra buffer being left un-trimmed when it could have 4011 // been. In general, this is not typical, and we would rather 4012 // optimized away the TS calculation below for the more common case 4013 // where PTSes are contiguous. 4014 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4015 } else { 4016 // We have no next buffer. Compute the PTS of the frame following 4017 // the last frame in this buffer by computing the duration of of 4018 // this frame in media time units and adding it to the PTS of the 4019 // buffer. 4020 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4021 / mCblk->frameSize; 4022 4023 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4024 &bufEnd)) { 4025 ALOGE("Failed to convert frame count of %lld to media time" 4026 " duration" " (scale factor %d/%u) in %s", 4027 frameCount, 4028 mMediaTimeToSampleTransform.a_to_b_numer, 4029 mMediaTimeToSampleTransform.a_to_b_denom, 4030 __PRETTY_FUNCTION__); 4031 break; 4032 } 4033 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4034 } 4035 4036 if (bufEnd > mediaTimeNow) 4037 break; 4038 4039 // Is the buffer we want to use in the middle of a mix operation right 4040 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4041 // from the mixer which should be coming back shortly. 4042 if (!trimEnd && mQueueHeadInFlight) { 4043 mTrimQueueHeadOnRelease = true; 4044 } 4045 } 4046 4047 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4048 if (trimStart < trimEnd) { 4049 // Update the bookkeeping for framesReady() 4050 for (size_t i = trimStart; i < trimEnd; ++i) { 4051 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4052 } 4053 4054 // Now actually remove the buffers from the queue. 4055 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4056 } 4057} 4058 4059void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4060 const char* logTag) { 4061 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4062 "%s called (reason \"%s\"), but timed buffer queue has no" 4063 " elements to trim.", __FUNCTION__, logTag); 4064 4065 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4066 mTimedBufferQueue.removeAt(0); 4067} 4068 4069void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4070 const TimedBuffer& buf, 4071 const char* logTag) { 4072 uint32_t bufBytes = buf.buffer()->size(); 4073 uint32_t consumedAlready = buf.position(); 4074 4075 ALOG_ASSERT(consumedAlready <= bufBytes, 4076 "Bad bookkeeping while updating frames pending. Timed buffer is" 4077 " only %u bytes long, but claims to have consumed %u" 4078 " bytes. (update reason: \"%s\")", 4079 bufBytes, consumedAlready, logTag); 4080 4081 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4082 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4083 "Bad bookkeeping while updating frames pending. Should have at" 4084 " least %u queued frames, but we think we have only %u. (update" 4085 " reason: \"%s\")", 4086 bufFrames, mFramesPendingInQueue, logTag); 4087 4088 mFramesPendingInQueue -= bufFrames; 4089} 4090 4091status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4092 const sp<IMemory>& buffer, int64_t pts) { 4093 4094 { 4095 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4096 if (!mMediaTimeTransformValid) 4097 return INVALID_OPERATION; 4098 } 4099 4100 Mutex::Autolock _l(mTimedBufferQueueLock); 4101 4102 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4103 mFramesPendingInQueue += bufFrames; 4104 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4105 4106 return NO_ERROR; 4107} 4108 4109status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4110 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4111 4112 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4113 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4114 target); 4115 4116 if (!(target == TimedAudioTrack::LOCAL_TIME || 4117 target == TimedAudioTrack::COMMON_TIME)) { 4118 return BAD_VALUE; 4119 } 4120 4121 Mutex::Autolock lock(mMediaTimeTransformLock); 4122 mMediaTimeTransform = xform; 4123 mMediaTimeTransformTarget = target; 4124 mMediaTimeTransformValid = true; 4125 4126 return NO_ERROR; 4127} 4128 4129#define min(a, b) ((a) < (b) ? (a) : (b)) 4130 4131// implementation of getNextBuffer for tracks whose buffers have timestamps 4132status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4133 AudioBufferProvider::Buffer* buffer, int64_t pts) 4134{ 4135 if (pts == AudioBufferProvider::kInvalidPTS) { 4136 buffer->raw = 0; 4137 buffer->frameCount = 0; 4138 mTimedAudioOutputOnTime = false; 4139 return INVALID_OPERATION; 4140 } 4141 4142 Mutex::Autolock _l(mTimedBufferQueueLock); 4143 4144 ALOG_ASSERT(!mQueueHeadInFlight, 4145 "getNextBuffer called without releaseBuffer!"); 4146 4147 while (true) { 4148 4149 // if we have no timed buffers, then fail 4150 if (mTimedBufferQueue.isEmpty()) { 4151 buffer->raw = 0; 4152 buffer->frameCount = 0; 4153 return NOT_ENOUGH_DATA; 4154 } 4155 4156 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4157 4158 // calculate the PTS of the head of the timed buffer queue expressed in 4159 // local time 4160 int64_t headLocalPTS; 4161 { 4162 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4163 4164 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4165 4166 if (mMediaTimeTransform.a_to_b_denom == 0) { 4167 // the transform represents a pause, so yield silence 4168 timedYieldSilence_l(buffer->frameCount, buffer); 4169 return NO_ERROR; 4170 } 4171 4172 int64_t transformedPTS; 4173 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4174 &transformedPTS)) { 4175 // the transform failed. this shouldn't happen, but if it does 4176 // then just drop this buffer 4177 ALOGW("timedGetNextBuffer transform failed"); 4178 buffer->raw = 0; 4179 buffer->frameCount = 0; 4180 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4181 return NO_ERROR; 4182 } 4183 4184 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4185 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4186 &headLocalPTS)) { 4187 buffer->raw = 0; 4188 buffer->frameCount = 0; 4189 return INVALID_OPERATION; 4190 } 4191 } else { 4192 headLocalPTS = transformedPTS; 4193 } 4194 } 4195 4196 // adjust the head buffer's PTS to reflect the portion of the head buffer 4197 // that has already been consumed 4198 int64_t effectivePTS = headLocalPTS + 4199 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4200 4201 // Calculate the delta in samples between the head of the input buffer 4202 // queue and the start of the next output buffer that will be written. 4203 // If the transformation fails because of over or underflow, it means 4204 // that the sample's position in the output stream is so far out of 4205 // whack that it should just be dropped. 4206 int64_t sampleDelta; 4207 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4208 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4209 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4210 " mix"); 4211 continue; 4212 } 4213 if (!mLocalTimeToSampleTransform.doForwardTransform( 4214 (effectivePTS - pts) << 32, &sampleDelta)) { 4215 ALOGV("*** too late during sample rate transform: dropped buffer"); 4216 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 4217 continue; 4218 } 4219 4220 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 4221 " sampleDelta=[%d.%08x]", 4222 head.pts(), head.position(), pts, 4223 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 4224 + (sampleDelta >> 32)), 4225 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4226 4227 // if the delta between the ideal placement for the next input sample and 4228 // the current output position is within this threshold, then we will 4229 // concatenate the next input samples to the previous output 4230 const int64_t kSampleContinuityThreshold = 4231 (static_cast<int64_t>(sampleRate()) << 32) / 250; 4232 4233 // if this is the first buffer of audio that we're emitting from this track 4234 // then it should be almost exactly on time. 4235 const int64_t kSampleStartupThreshold = 1LL << 32; 4236 4237 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4238 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4239 // the next input is close enough to being on time, so concatenate it 4240 // with the last output 4241 timedYieldSamples_l(buffer); 4242 4243 ALOGVV("*** on time: head.pos=%d frameCount=%u", 4244 head.position(), buffer->frameCount); 4245 return NO_ERROR; 4246 } 4247 4248 // Looks like our output is not on time. Reset our on timed status. 4249 // Next time we mix samples from our input queue, then should be within 4250 // the StartupThreshold. 4251 mTimedAudioOutputOnTime = false; 4252 if (sampleDelta > 0) { 4253 // the gap between the current output position and the proper start of 4254 // the next input sample is too big, so fill it with silence 4255 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4256 4257 timedYieldSilence_l(framesUntilNextInput, buffer); 4258 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4259 return NO_ERROR; 4260 } else { 4261 // the next input sample is late 4262 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4263 size_t onTimeSamplePosition = 4264 head.position() + lateFrames * mCblk->frameSize; 4265 4266 if (onTimeSamplePosition > head.buffer()->size()) { 4267 // all the remaining samples in the head are too late, so 4268 // drop it and move on 4269 ALOGV("*** too late: dropped buffer"); 4270 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 4271 continue; 4272 } else { 4273 // skip over the late samples 4274 head.setPosition(onTimeSamplePosition); 4275 4276 // yield the available samples 4277 timedYieldSamples_l(buffer); 4278 4279 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4280 return NO_ERROR; 4281 } 4282 } 4283 } 4284} 4285 4286// Yield samples from the timed buffer queue head up to the given output 4287// buffer's capacity. 4288// 4289// Caller must hold mTimedBufferQueueLock 4290void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 4291 AudioBufferProvider::Buffer* buffer) { 4292 4293 const TimedBuffer& head = mTimedBufferQueue[0]; 4294 4295 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4296 head.position()); 4297 4298 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4299 mCblk->frameSize); 4300 size_t framesRequested = buffer->frameCount; 4301 buffer->frameCount = min(framesLeftInHead, framesRequested); 4302 4303 mQueueHeadInFlight = true; 4304 mTimedAudioOutputOnTime = true; 4305} 4306 4307// Yield samples of silence up to the given output buffer's capacity 4308// 4309// Caller must hold mTimedBufferQueueLock 4310void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 4311 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4312 4313 // lazily allocate a buffer filled with silence 4314 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4315 delete [] mTimedSilenceBuffer; 4316 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4317 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4318 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4319 } 4320 4321 buffer->raw = mTimedSilenceBuffer; 4322 size_t framesRequested = buffer->frameCount; 4323 buffer->frameCount = min(numFrames, framesRequested); 4324 4325 mTimedAudioOutputOnTime = false; 4326} 4327 4328// AudioBufferProvider interface 4329void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4330 AudioBufferProvider::Buffer* buffer) { 4331 4332 Mutex::Autolock _l(mTimedBufferQueueLock); 4333 4334 // If the buffer which was just released is part of the buffer at the head 4335 // of the queue, be sure to update the amt of the buffer which has been 4336 // consumed. If the buffer being returned is not part of the head of the 4337 // queue, its either because the buffer is part of the silence buffer, or 4338 // because the head of the timed queue was trimmed after the mixer called 4339 // getNextBuffer but before the mixer called releaseBuffer. 4340 if (buffer->raw == mTimedSilenceBuffer) { 4341 ALOG_ASSERT(!mQueueHeadInFlight, 4342 "Queue head in flight during release of silence buffer!"); 4343 goto done; 4344 } 4345 4346 ALOG_ASSERT(mQueueHeadInFlight, 4347 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 4348 " head in flight."); 4349 4350 if (mTimedBufferQueue.size()) { 4351 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4352 4353 void* start = head.buffer()->pointer(); 4354 void* end = reinterpret_cast<void*>( 4355 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 4356 + head.buffer()->size()); 4357 4358 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 4359 "released buffer not within the head of the timed buffer" 4360 " queue; qHead = [%p, %p], released buffer = %p", 4361 start, end, buffer->raw); 4362 4363 head.setPosition(head.position() + 4364 (buffer->frameCount * mCblk->frameSize)); 4365 mQueueHeadInFlight = false; 4366 4367 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 4368 "Bad bookkeeping during releaseBuffer! Should have at" 4369 " least %u queued frames, but we think we have only %u", 4370 buffer->frameCount, mFramesPendingInQueue); 4371 4372 mFramesPendingInQueue -= buffer->frameCount; 4373 4374 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 4375 || mTrimQueueHeadOnRelease) { 4376 trimTimedBufferQueueHead_l("releaseBuffer"); 4377 mTrimQueueHeadOnRelease = false; 4378 } 4379 } else { 4380 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 4381 " buffers in the timed buffer queue"); 4382 } 4383 4384done: 4385 buffer->raw = 0; 4386 buffer->frameCount = 0; 4387} 4388 4389uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4390 Mutex::Autolock _l(mTimedBufferQueueLock); 4391 return mFramesPendingInQueue; 4392} 4393 4394AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4395 : mPTS(0), mPosition(0) {} 4396 4397AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4398 const sp<IMemory>& buffer, int64_t pts) 4399 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4400 4401// ---------------------------------------------------------------------------- 4402 4403// RecordTrack constructor must be called with AudioFlinger::mLock held 4404AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4405 RecordThread *thread, 4406 const sp<Client>& client, 4407 uint32_t sampleRate, 4408 audio_format_t format, 4409 uint32_t channelMask, 4410 int frameCount, 4411 int sessionId) 4412 : TrackBase(thread, client, sampleRate, format, 4413 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4414 mOverflow(false) 4415{ 4416 if (mCblk != NULL) { 4417 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4418 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4419 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4420 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4421 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4422 } else { 4423 mCblk->frameSize = sizeof(int8_t); 4424 } 4425 } 4426} 4427 4428AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4429{ 4430 sp<ThreadBase> thread = mThread.promote(); 4431 if (thread != 0) { 4432 AudioSystem::releaseInput(thread->id()); 4433 } 4434} 4435 4436// AudioBufferProvider interface 4437status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4438{ 4439 audio_track_cblk_t* cblk = this->cblk(); 4440 uint32_t framesAvail; 4441 uint32_t framesReq = buffer->frameCount; 4442 4443 // Check if last stepServer failed, try to step now 4444 if (mStepServerFailed) { 4445 if (!step()) goto getNextBuffer_exit; 4446 ALOGV("stepServer recovered"); 4447 mStepServerFailed = false; 4448 } 4449 4450 framesAvail = cblk->framesAvailable_l(); 4451 4452 if (CC_LIKELY(framesAvail)) { 4453 uint32_t s = cblk->server; 4454 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4455 4456 if (framesReq > framesAvail) { 4457 framesReq = framesAvail; 4458 } 4459 if (framesReq > bufferEnd - s) { 4460 framesReq = bufferEnd - s; 4461 } 4462 4463 buffer->raw = getBuffer(s, framesReq); 4464 if (buffer->raw == NULL) goto getNextBuffer_exit; 4465 4466 buffer->frameCount = framesReq; 4467 return NO_ERROR; 4468 } 4469 4470getNextBuffer_exit: 4471 buffer->raw = NULL; 4472 buffer->frameCount = 0; 4473 return NOT_ENOUGH_DATA; 4474} 4475 4476status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 4477 int triggerSession) 4478{ 4479 sp<ThreadBase> thread = mThread.promote(); 4480 if (thread != 0) { 4481 RecordThread *recordThread = (RecordThread *)thread.get(); 4482 return recordThread->start(this, event, triggerSession); 4483 } else { 4484 return BAD_VALUE; 4485 } 4486} 4487 4488void AudioFlinger::RecordThread::RecordTrack::stop() 4489{ 4490 sp<ThreadBase> thread = mThread.promote(); 4491 if (thread != 0) { 4492 RecordThread *recordThread = (RecordThread *)thread.get(); 4493 recordThread->stop(this); 4494 TrackBase::reset(); 4495 // Force overrun condition to avoid false overrun callback until first data is 4496 // read from buffer 4497 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4498 } 4499} 4500 4501void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4502{ 4503 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4504 (mClient == 0) ? getpid_cached : mClient->pid(), 4505 mFormat, 4506 mChannelMask, 4507 mSessionId, 4508 mFrameCount, 4509 mState, 4510 mCblk->sampleRate, 4511 mCblk->server, 4512 mCblk->user); 4513} 4514 4515 4516// ---------------------------------------------------------------------------- 4517 4518AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4519 PlaybackThread *playbackThread, 4520 DuplicatingThread *sourceThread, 4521 uint32_t sampleRate, 4522 audio_format_t format, 4523 uint32_t channelMask, 4524 int frameCount) 4525 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 4526 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 4527 mActive(false), mSourceThread(sourceThread) 4528{ 4529 4530 if (mCblk != NULL) { 4531 mCblk->flags |= CBLK_DIRECTION_OUT; 4532 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4533 mOutBuffer.frameCount = 0; 4534 playbackThread->mTracks.add(this); 4535 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4536 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4537 mCblk, mBuffer, mCblk->buffers, 4538 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4539 } else { 4540 ALOGW("Error creating output track on thread %p", playbackThread); 4541 } 4542} 4543 4544AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4545{ 4546 clearBufferQueue(); 4547} 4548 4549status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 4550 int triggerSession) 4551{ 4552 status_t status = Track::start(event, triggerSession); 4553 if (status != NO_ERROR) { 4554 return status; 4555 } 4556 4557 mActive = true; 4558 mRetryCount = 127; 4559 return status; 4560} 4561 4562void AudioFlinger::PlaybackThread::OutputTrack::stop() 4563{ 4564 Track::stop(); 4565 clearBufferQueue(); 4566 mOutBuffer.frameCount = 0; 4567 mActive = false; 4568} 4569 4570bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4571{ 4572 Buffer *pInBuffer; 4573 Buffer inBuffer; 4574 uint32_t channelCount = mChannelCount; 4575 bool outputBufferFull = false; 4576 inBuffer.frameCount = frames; 4577 inBuffer.i16 = data; 4578 4579 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4580 4581 if (!mActive && frames != 0) { 4582 start(); 4583 sp<ThreadBase> thread = mThread.promote(); 4584 if (thread != 0) { 4585 MixerThread *mixerThread = (MixerThread *)thread.get(); 4586 if (mCblk->frameCount > frames){ 4587 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4588 uint32_t startFrames = (mCblk->frameCount - frames); 4589 pInBuffer = new Buffer; 4590 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4591 pInBuffer->frameCount = startFrames; 4592 pInBuffer->i16 = pInBuffer->mBuffer; 4593 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4594 mBufferQueue.add(pInBuffer); 4595 } else { 4596 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4597 } 4598 } 4599 } 4600 } 4601 4602 while (waitTimeLeftMs) { 4603 // First write pending buffers, then new data 4604 if (mBufferQueue.size()) { 4605 pInBuffer = mBufferQueue.itemAt(0); 4606 } else { 4607 pInBuffer = &inBuffer; 4608 } 4609 4610 if (pInBuffer->frameCount == 0) { 4611 break; 4612 } 4613 4614 if (mOutBuffer.frameCount == 0) { 4615 mOutBuffer.frameCount = pInBuffer->frameCount; 4616 nsecs_t startTime = systemTime(); 4617 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4618 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4619 outputBufferFull = true; 4620 break; 4621 } 4622 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4623 if (waitTimeLeftMs >= waitTimeMs) { 4624 waitTimeLeftMs -= waitTimeMs; 4625 } else { 4626 waitTimeLeftMs = 0; 4627 } 4628 } 4629 4630 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4631 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4632 mCblk->stepUser(outFrames); 4633 pInBuffer->frameCount -= outFrames; 4634 pInBuffer->i16 += outFrames * channelCount; 4635 mOutBuffer.frameCount -= outFrames; 4636 mOutBuffer.i16 += outFrames * channelCount; 4637 4638 if (pInBuffer->frameCount == 0) { 4639 if (mBufferQueue.size()) { 4640 mBufferQueue.removeAt(0); 4641 delete [] pInBuffer->mBuffer; 4642 delete pInBuffer; 4643 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4644 } else { 4645 break; 4646 } 4647 } 4648 } 4649 4650 // If we could not write all frames, allocate a buffer and queue it for next time. 4651 if (inBuffer.frameCount) { 4652 sp<ThreadBase> thread = mThread.promote(); 4653 if (thread != 0 && !thread->standby()) { 4654 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4655 pInBuffer = new Buffer; 4656 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4657 pInBuffer->frameCount = inBuffer.frameCount; 4658 pInBuffer->i16 = pInBuffer->mBuffer; 4659 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4660 mBufferQueue.add(pInBuffer); 4661 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4662 } else { 4663 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4664 } 4665 } 4666 } 4667 4668 // Calling write() with a 0 length buffer, means that no more data will be written: 4669 // If no more buffers are pending, fill output track buffer to make sure it is started 4670 // by output mixer. 4671 if (frames == 0 && mBufferQueue.size() == 0) { 4672 if (mCblk->user < mCblk->frameCount) { 4673 frames = mCblk->frameCount - mCblk->user; 4674 pInBuffer = new Buffer; 4675 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4676 pInBuffer->frameCount = frames; 4677 pInBuffer->i16 = pInBuffer->mBuffer; 4678 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4679 mBufferQueue.add(pInBuffer); 4680 } else if (mActive) { 4681 stop(); 4682 } 4683 } 4684 4685 return outputBufferFull; 4686} 4687 4688status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4689{ 4690 int active; 4691 status_t result; 4692 audio_track_cblk_t* cblk = mCblk; 4693 uint32_t framesReq = buffer->frameCount; 4694 4695// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4696 buffer->frameCount = 0; 4697 4698 uint32_t framesAvail = cblk->framesAvailable(); 4699 4700 4701 if (framesAvail == 0) { 4702 Mutex::Autolock _l(cblk->lock); 4703 goto start_loop_here; 4704 while (framesAvail == 0) { 4705 active = mActive; 4706 if (CC_UNLIKELY(!active)) { 4707 ALOGV("Not active and NO_MORE_BUFFERS"); 4708 return NO_MORE_BUFFERS; 4709 } 4710 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4711 if (result != NO_ERROR) { 4712 return NO_MORE_BUFFERS; 4713 } 4714 // read the server count again 4715 start_loop_here: 4716 framesAvail = cblk->framesAvailable_l(); 4717 } 4718 } 4719 4720// if (framesAvail < framesReq) { 4721// return NO_MORE_BUFFERS; 4722// } 4723 4724 if (framesReq > framesAvail) { 4725 framesReq = framesAvail; 4726 } 4727 4728 uint32_t u = cblk->user; 4729 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4730 4731 if (framesReq > bufferEnd - u) { 4732 framesReq = bufferEnd - u; 4733 } 4734 4735 buffer->frameCount = framesReq; 4736 buffer->raw = (void *)cblk->buffer(u); 4737 return NO_ERROR; 4738} 4739 4740 4741void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4742{ 4743 size_t size = mBufferQueue.size(); 4744 4745 for (size_t i = 0; i < size; i++) { 4746 Buffer *pBuffer = mBufferQueue.itemAt(i); 4747 delete [] pBuffer->mBuffer; 4748 delete pBuffer; 4749 } 4750 mBufferQueue.clear(); 4751} 4752 4753// ---------------------------------------------------------------------------- 4754 4755AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4756 : RefBase(), 4757 mAudioFlinger(audioFlinger), 4758 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4759 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4760 mPid(pid), 4761 mTimedTrackCount(0) 4762{ 4763 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4764} 4765 4766// Client destructor must be called with AudioFlinger::mLock held 4767AudioFlinger::Client::~Client() 4768{ 4769 mAudioFlinger->removeClient_l(mPid); 4770} 4771 4772sp<MemoryDealer> AudioFlinger::Client::heap() const 4773{ 4774 return mMemoryDealer; 4775} 4776 4777// Reserve one of the limited slots for a timed audio track associated 4778// with this client 4779bool AudioFlinger::Client::reserveTimedTrack() 4780{ 4781 const int kMaxTimedTracksPerClient = 4; 4782 4783 Mutex::Autolock _l(mTimedTrackLock); 4784 4785 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4786 ALOGW("can not create timed track - pid %d has exceeded the limit", 4787 mPid); 4788 return false; 4789 } 4790 4791 mTimedTrackCount++; 4792 return true; 4793} 4794 4795// Release a slot for a timed audio track 4796void AudioFlinger::Client::releaseTimedTrack() 4797{ 4798 Mutex::Autolock _l(mTimedTrackLock); 4799 mTimedTrackCount--; 4800} 4801 4802// ---------------------------------------------------------------------------- 4803 4804AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4805 const sp<IAudioFlingerClient>& client, 4806 pid_t pid) 4807 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4808{ 4809} 4810 4811AudioFlinger::NotificationClient::~NotificationClient() 4812{ 4813} 4814 4815void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4816{ 4817 sp<NotificationClient> keep(this); 4818 mAudioFlinger->removeNotificationClient(mPid); 4819} 4820 4821// ---------------------------------------------------------------------------- 4822 4823AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4824 : BnAudioTrack(), 4825 mTrack(track) 4826{ 4827} 4828 4829AudioFlinger::TrackHandle::~TrackHandle() { 4830 // just stop the track on deletion, associated resources 4831 // will be freed from the main thread once all pending buffers have 4832 // been played. Unless it's not in the active track list, in which 4833 // case we free everything now... 4834 mTrack->destroy(); 4835} 4836 4837sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4838 return mTrack->getCblk(); 4839} 4840 4841status_t AudioFlinger::TrackHandle::start() { 4842 return mTrack->start(); 4843} 4844 4845void AudioFlinger::TrackHandle::stop() { 4846 mTrack->stop(); 4847} 4848 4849void AudioFlinger::TrackHandle::flush() { 4850 mTrack->flush(); 4851} 4852 4853void AudioFlinger::TrackHandle::mute(bool e) { 4854 mTrack->mute(e); 4855} 4856 4857void AudioFlinger::TrackHandle::pause() { 4858 mTrack->pause(); 4859} 4860 4861status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4862{ 4863 return mTrack->attachAuxEffect(EffectId); 4864} 4865 4866status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4867 sp<IMemory>* buffer) { 4868 if (!mTrack->isTimedTrack()) 4869 return INVALID_OPERATION; 4870 4871 PlaybackThread::TimedTrack* tt = 4872 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4873 return tt->allocateTimedBuffer(size, buffer); 4874} 4875 4876status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4877 int64_t pts) { 4878 if (!mTrack->isTimedTrack()) 4879 return INVALID_OPERATION; 4880 4881 PlaybackThread::TimedTrack* tt = 4882 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4883 return tt->queueTimedBuffer(buffer, pts); 4884} 4885 4886status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4887 const LinearTransform& xform, int target) { 4888 4889 if (!mTrack->isTimedTrack()) 4890 return INVALID_OPERATION; 4891 4892 PlaybackThread::TimedTrack* tt = 4893 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4894 return tt->setMediaTimeTransform( 4895 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4896} 4897 4898status_t AudioFlinger::TrackHandle::onTransact( 4899 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4900{ 4901 return BnAudioTrack::onTransact(code, data, reply, flags); 4902} 4903 4904// ---------------------------------------------------------------------------- 4905 4906sp<IAudioRecord> AudioFlinger::openRecord( 4907 pid_t pid, 4908 audio_io_handle_t input, 4909 uint32_t sampleRate, 4910 audio_format_t format, 4911 uint32_t channelMask, 4912 int frameCount, 4913 IAudioFlinger::track_flags_t flags, 4914 int *sessionId, 4915 status_t *status) 4916{ 4917 sp<RecordThread::RecordTrack> recordTrack; 4918 sp<RecordHandle> recordHandle; 4919 sp<Client> client; 4920 status_t lStatus; 4921 RecordThread *thread; 4922 size_t inFrameCount; 4923 int lSessionId; 4924 4925 // check calling permissions 4926 if (!recordingAllowed()) { 4927 lStatus = PERMISSION_DENIED; 4928 goto Exit; 4929 } 4930 4931 // add client to list 4932 { // scope for mLock 4933 Mutex::Autolock _l(mLock); 4934 thread = checkRecordThread_l(input); 4935 if (thread == NULL) { 4936 lStatus = BAD_VALUE; 4937 goto Exit; 4938 } 4939 4940 client = registerPid_l(pid); 4941 4942 // If no audio session id is provided, create one here 4943 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4944 lSessionId = *sessionId; 4945 } else { 4946 lSessionId = nextUniqueId(); 4947 if (sessionId != NULL) { 4948 *sessionId = lSessionId; 4949 } 4950 } 4951 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4952 recordTrack = thread->createRecordTrack_l(client, 4953 sampleRate, 4954 format, 4955 channelMask, 4956 frameCount, 4957 lSessionId, 4958 &lStatus); 4959 } 4960 if (lStatus != NO_ERROR) { 4961 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4962 // destructor is called by the TrackBase destructor with mLock held 4963 client.clear(); 4964 recordTrack.clear(); 4965 goto Exit; 4966 } 4967 4968 // return to handle to client 4969 recordHandle = new RecordHandle(recordTrack); 4970 lStatus = NO_ERROR; 4971 4972Exit: 4973 if (status) { 4974 *status = lStatus; 4975 } 4976 return recordHandle; 4977} 4978 4979// ---------------------------------------------------------------------------- 4980 4981AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4982 : BnAudioRecord(), 4983 mRecordTrack(recordTrack) 4984{ 4985} 4986 4987AudioFlinger::RecordHandle::~RecordHandle() { 4988 stop(); 4989} 4990 4991sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4992 return mRecordTrack->getCblk(); 4993} 4994 4995status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 4996 ALOGV("RecordHandle::start()"); 4997 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 4998} 4999 5000void AudioFlinger::RecordHandle::stop() { 5001 ALOGV("RecordHandle::stop()"); 5002 mRecordTrack->stop(); 5003} 5004 5005status_t AudioFlinger::RecordHandle::onTransact( 5006 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5007{ 5008 return BnAudioRecord::onTransact(code, data, reply, flags); 5009} 5010 5011// ---------------------------------------------------------------------------- 5012 5013AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5014 AudioStreamIn *input, 5015 uint32_t sampleRate, 5016 uint32_t channels, 5017 audio_io_handle_t id, 5018 uint32_t device) : 5019 ThreadBase(audioFlinger, id, device, RECORD), 5020 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5021 // mRsmpInIndex and mInputBytes set by readInputParameters() 5022 mReqChannelCount(popcount(channels)), 5023 mReqSampleRate(sampleRate) 5024 // mBytesRead is only meaningful while active, and so is cleared in start() 5025 // (but might be better to also clear here for dump?) 5026{ 5027 snprintf(mName, kNameLength, "AudioIn_%X", id); 5028 5029 readInputParameters(); 5030} 5031 5032 5033AudioFlinger::RecordThread::~RecordThread() 5034{ 5035 delete[] mRsmpInBuffer; 5036 delete mResampler; 5037 delete[] mRsmpOutBuffer; 5038} 5039 5040void AudioFlinger::RecordThread::onFirstRef() 5041{ 5042 run(mName, PRIORITY_URGENT_AUDIO); 5043} 5044 5045status_t AudioFlinger::RecordThread::readyToRun() 5046{ 5047 status_t status = initCheck(); 5048 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5049 return status; 5050} 5051 5052bool AudioFlinger::RecordThread::threadLoop() 5053{ 5054 AudioBufferProvider::Buffer buffer; 5055 sp<RecordTrack> activeTrack; 5056 Vector< sp<EffectChain> > effectChains; 5057 5058 nsecs_t lastWarning = 0; 5059 5060 acquireWakeLock(); 5061 5062 // start recording 5063 while (!exitPending()) { 5064 5065 processConfigEvents(); 5066 5067 { // scope for mLock 5068 Mutex::Autolock _l(mLock); 5069 checkForNewParameters_l(); 5070 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5071 if (!mStandby) { 5072 mInput->stream->common.standby(&mInput->stream->common); 5073 mStandby = true; 5074 } 5075 5076 if (exitPending()) break; 5077 5078 releaseWakeLock_l(); 5079 ALOGV("RecordThread: loop stopping"); 5080 // go to sleep 5081 mWaitWorkCV.wait(mLock); 5082 ALOGV("RecordThread: loop starting"); 5083 acquireWakeLock_l(); 5084 continue; 5085 } 5086 if (mActiveTrack != 0) { 5087 if (mActiveTrack->mState == TrackBase::PAUSING) { 5088 if (!mStandby) { 5089 mInput->stream->common.standby(&mInput->stream->common); 5090 mStandby = true; 5091 } 5092 mActiveTrack.clear(); 5093 mStartStopCond.broadcast(); 5094 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5095 if (mReqChannelCount != mActiveTrack->channelCount()) { 5096 mActiveTrack.clear(); 5097 mStartStopCond.broadcast(); 5098 } else if (mBytesRead != 0) { 5099 // record start succeeds only if first read from audio input 5100 // succeeds 5101 if (mBytesRead > 0) { 5102 mActiveTrack->mState = TrackBase::ACTIVE; 5103 } else { 5104 mActiveTrack.clear(); 5105 } 5106 mStartStopCond.broadcast(); 5107 } 5108 mStandby = false; 5109 } 5110 } 5111 lockEffectChains_l(effectChains); 5112 } 5113 5114 if (mActiveTrack != 0) { 5115 if (mActiveTrack->mState != TrackBase::ACTIVE && 5116 mActiveTrack->mState != TrackBase::RESUMING) { 5117 unlockEffectChains(effectChains); 5118 usleep(kRecordThreadSleepUs); 5119 continue; 5120 } 5121 for (size_t i = 0; i < effectChains.size(); i ++) { 5122 effectChains[i]->process_l(); 5123 } 5124 5125 buffer.frameCount = mFrameCount; 5126 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5127 size_t framesOut = buffer.frameCount; 5128 if (mResampler == NULL) { 5129 // no resampling 5130 while (framesOut) { 5131 size_t framesIn = mFrameCount - mRsmpInIndex; 5132 if (framesIn) { 5133 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5134 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5135 if (framesIn > framesOut) 5136 framesIn = framesOut; 5137 mRsmpInIndex += framesIn; 5138 framesOut -= framesIn; 5139 if ((int)mChannelCount == mReqChannelCount || 5140 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5141 memcpy(dst, src, framesIn * mFrameSize); 5142 } else { 5143 int16_t *src16 = (int16_t *)src; 5144 int16_t *dst16 = (int16_t *)dst; 5145 if (mChannelCount == 1) { 5146 while (framesIn--) { 5147 *dst16++ = *src16; 5148 *dst16++ = *src16++; 5149 } 5150 } else { 5151 while (framesIn--) { 5152 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5153 src16 += 2; 5154 } 5155 } 5156 } 5157 } 5158 if (framesOut && mFrameCount == mRsmpInIndex) { 5159 if (framesOut == mFrameCount && 5160 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5161 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5162 framesOut = 0; 5163 } else { 5164 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5165 mRsmpInIndex = 0; 5166 } 5167 if (mBytesRead < 0) { 5168 ALOGE("Error reading audio input"); 5169 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5170 // Force input into standby so that it tries to 5171 // recover at next read attempt 5172 mInput->stream->common.standby(&mInput->stream->common); 5173 usleep(kRecordThreadSleepUs); 5174 } 5175 mRsmpInIndex = mFrameCount; 5176 framesOut = 0; 5177 buffer.frameCount = 0; 5178 } 5179 } 5180 } 5181 } else { 5182 // resampling 5183 5184 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5185 // alter output frame count as if we were expecting stereo samples 5186 if (mChannelCount == 1 && mReqChannelCount == 1) { 5187 framesOut >>= 1; 5188 } 5189 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5190 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5191 // are 32 bit aligned which should be always true. 5192 if (mChannelCount == 2 && mReqChannelCount == 1) { 5193 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5194 // the resampler always outputs stereo samples: do post stereo to mono conversion 5195 int16_t *src = (int16_t *)mRsmpOutBuffer; 5196 int16_t *dst = buffer.i16; 5197 while (framesOut--) { 5198 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5199 src += 2; 5200 } 5201 } else { 5202 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5203 } 5204 5205 } 5206 if (mFramestoDrop == 0) { 5207 mActiveTrack->releaseBuffer(&buffer); 5208 } else { 5209 if (mFramestoDrop > 0) { 5210 mFramestoDrop -= buffer.frameCount; 5211 if (mFramestoDrop < 0) { 5212 mFramestoDrop = 0; 5213 } 5214 } 5215 } 5216 mActiveTrack->overflow(); 5217 } 5218 // client isn't retrieving buffers fast enough 5219 else { 5220 if (!mActiveTrack->setOverflow()) { 5221 nsecs_t now = systemTime(); 5222 if ((now - lastWarning) > kWarningThrottleNs) { 5223 ALOGW("RecordThread: buffer overflow"); 5224 lastWarning = now; 5225 } 5226 } 5227 // Release the processor for a while before asking for a new buffer. 5228 // This will give the application more chance to read from the buffer and 5229 // clear the overflow. 5230 usleep(kRecordThreadSleepUs); 5231 } 5232 } 5233 // enable changes in effect chain 5234 unlockEffectChains(effectChains); 5235 effectChains.clear(); 5236 } 5237 5238 if (!mStandby) { 5239 mInput->stream->common.standby(&mInput->stream->common); 5240 } 5241 mActiveTrack.clear(); 5242 5243 mStartStopCond.broadcast(); 5244 5245 releaseWakeLock(); 5246 5247 ALOGV("RecordThread %p exiting", this); 5248 return false; 5249} 5250 5251 5252sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5253 const sp<AudioFlinger::Client>& client, 5254 uint32_t sampleRate, 5255 audio_format_t format, 5256 int channelMask, 5257 int frameCount, 5258 int sessionId, 5259 status_t *status) 5260{ 5261 sp<RecordTrack> track; 5262 status_t lStatus; 5263 5264 lStatus = initCheck(); 5265 if (lStatus != NO_ERROR) { 5266 ALOGE("Audio driver not initialized."); 5267 goto Exit; 5268 } 5269 5270 { // scope for mLock 5271 Mutex::Autolock _l(mLock); 5272 5273 track = new RecordTrack(this, client, sampleRate, 5274 format, channelMask, frameCount, sessionId); 5275 5276 if (track->getCblk() == 0) { 5277 lStatus = NO_MEMORY; 5278 goto Exit; 5279 } 5280 5281 mTrack = track.get(); 5282 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5283 bool suspend = audio_is_bluetooth_sco_device( 5284 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5285 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5286 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5287 } 5288 lStatus = NO_ERROR; 5289 5290Exit: 5291 if (status) { 5292 *status = lStatus; 5293 } 5294 return track; 5295} 5296 5297status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5298 AudioSystem::sync_event_t event, 5299 int triggerSession) 5300{ 5301 // FIXME use tid here 5302 ALOGV("RecordThread::start tid=%d, event %d, triggerSession %d", tid, event, triggerSession); 5303 sp<ThreadBase> strongMe = this; 5304 status_t status = NO_ERROR; 5305 5306 if (event == AudioSystem::SYNC_EVENT_NONE) { 5307 mSyncStartEvent.clear(); 5308 mFramestoDrop = 0; 5309 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5310 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5311 triggerSession, 5312 recordTrack->sessionId(), 5313 syncStartEventCallback, 5314 this); 5315 mFramestoDrop = -1; 5316 } 5317 5318 { 5319 AutoMutex lock(mLock); 5320 if (mActiveTrack != 0) { 5321 if (recordTrack != mActiveTrack.get()) { 5322 status = -EBUSY; 5323 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5324 mActiveTrack->mState = TrackBase::ACTIVE; 5325 } 5326 return status; 5327 } 5328 5329 recordTrack->mState = TrackBase::IDLE; 5330 mActiveTrack = recordTrack; 5331 mLock.unlock(); 5332 status_t status = AudioSystem::startInput(mId); 5333 mLock.lock(); 5334 if (status != NO_ERROR) { 5335 mActiveTrack.clear(); 5336 clearSyncStartEvent(); 5337 return status; 5338 } 5339 mRsmpInIndex = mFrameCount; 5340 mBytesRead = 0; 5341 if (mResampler != NULL) { 5342 mResampler->reset(); 5343 } 5344 mActiveTrack->mState = TrackBase::RESUMING; 5345 // signal thread to start 5346 ALOGV("Signal record thread"); 5347 mWaitWorkCV.signal(); 5348 // do not wait for mStartStopCond if exiting 5349 if (exitPending()) { 5350 mActiveTrack.clear(); 5351 status = INVALID_OPERATION; 5352 goto startError; 5353 } 5354 mStartStopCond.wait(mLock); 5355 if (mActiveTrack == 0) { 5356 ALOGV("Record failed to start"); 5357 status = BAD_VALUE; 5358 goto startError; 5359 } 5360 ALOGV("Record started OK"); 5361 return status; 5362 } 5363startError: 5364 AudioSystem::stopInput(mId); 5365 clearSyncStartEvent(); 5366 return status; 5367} 5368 5369void AudioFlinger::RecordThread::clearSyncStartEvent() 5370{ 5371 if (mSyncStartEvent != 0) { 5372 mSyncStartEvent->cancel(); 5373 } 5374 mSyncStartEvent.clear(); 5375} 5376 5377void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5378{ 5379 sp<SyncEvent> strongEvent = event.promote(); 5380 5381 if (strongEvent != 0) { 5382 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5383 me->handleSyncStartEvent(strongEvent); 5384 } 5385} 5386 5387void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5388{ 5389 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5390 mActiveTrack.get(), 5391 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5392 event->listenerSession()); 5393 5394 if (mActiveTrack != 0 && 5395 event == mSyncStartEvent) { 5396 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5397 // from audio HAL 5398 mFramestoDrop = mFrameCount * 2; 5399 mSyncStartEvent.clear(); 5400 } 5401} 5402 5403void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5404 ALOGV("RecordThread::stop"); 5405 sp<ThreadBase> strongMe = this; 5406 { 5407 AutoMutex lock(mLock); 5408 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5409 mActiveTrack->mState = TrackBase::PAUSING; 5410 // do not wait for mStartStopCond if exiting 5411 if (exitPending()) { 5412 return; 5413 } 5414 mStartStopCond.wait(mLock); 5415 // if we have been restarted, recordTrack == mActiveTrack.get() here 5416 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5417 mLock.unlock(); 5418 AudioSystem::stopInput(mId); 5419 mLock.lock(); 5420 ALOGV("Record stopped OK"); 5421 } 5422 } 5423 } 5424} 5425 5426bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5427{ 5428 return false; 5429} 5430 5431status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5432{ 5433 if (!isValidSyncEvent(event)) { 5434 return BAD_VALUE; 5435 } 5436 5437 Mutex::Autolock _l(mLock); 5438 5439 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5440 mTrack->setSyncEvent(event); 5441 return NO_ERROR; 5442 } 5443 return NAME_NOT_FOUND; 5444} 5445 5446status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5447{ 5448 const size_t SIZE = 256; 5449 char buffer[SIZE]; 5450 String8 result; 5451 5452 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5453 result.append(buffer); 5454 5455 if (mActiveTrack != 0) { 5456 result.append("Active Track:\n"); 5457 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5458 mActiveTrack->dump(buffer, SIZE); 5459 result.append(buffer); 5460 5461 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5462 result.append(buffer); 5463 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5464 result.append(buffer); 5465 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5466 result.append(buffer); 5467 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5468 result.append(buffer); 5469 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5470 result.append(buffer); 5471 5472 5473 } else { 5474 result.append("No record client\n"); 5475 } 5476 write(fd, result.string(), result.size()); 5477 5478 dumpBase(fd, args); 5479 dumpEffectChains(fd, args); 5480 5481 return NO_ERROR; 5482} 5483 5484// AudioBufferProvider interface 5485status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5486{ 5487 size_t framesReq = buffer->frameCount; 5488 size_t framesReady = mFrameCount - mRsmpInIndex; 5489 int channelCount; 5490 5491 if (framesReady == 0) { 5492 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5493 if (mBytesRead < 0) { 5494 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5495 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5496 // Force input into standby so that it tries to 5497 // recover at next read attempt 5498 mInput->stream->common.standby(&mInput->stream->common); 5499 usleep(kRecordThreadSleepUs); 5500 } 5501 buffer->raw = NULL; 5502 buffer->frameCount = 0; 5503 return NOT_ENOUGH_DATA; 5504 } 5505 mRsmpInIndex = 0; 5506 framesReady = mFrameCount; 5507 } 5508 5509 if (framesReq > framesReady) { 5510 framesReq = framesReady; 5511 } 5512 5513 if (mChannelCount == 1 && mReqChannelCount == 2) { 5514 channelCount = 1; 5515 } else { 5516 channelCount = 2; 5517 } 5518 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5519 buffer->frameCount = framesReq; 5520 return NO_ERROR; 5521} 5522 5523// AudioBufferProvider interface 5524void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5525{ 5526 mRsmpInIndex += buffer->frameCount; 5527 buffer->frameCount = 0; 5528} 5529 5530bool AudioFlinger::RecordThread::checkForNewParameters_l() 5531{ 5532 bool reconfig = false; 5533 5534 while (!mNewParameters.isEmpty()) { 5535 status_t status = NO_ERROR; 5536 String8 keyValuePair = mNewParameters[0]; 5537 AudioParameter param = AudioParameter(keyValuePair); 5538 int value; 5539 audio_format_t reqFormat = mFormat; 5540 int reqSamplingRate = mReqSampleRate; 5541 int reqChannelCount = mReqChannelCount; 5542 5543 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5544 reqSamplingRate = value; 5545 reconfig = true; 5546 } 5547 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5548 reqFormat = (audio_format_t) value; 5549 reconfig = true; 5550 } 5551 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5552 reqChannelCount = popcount(value); 5553 reconfig = true; 5554 } 5555 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5556 // do not accept frame count changes if tracks are open as the track buffer 5557 // size depends on frame count and correct behavior would not be guaranteed 5558 // if frame count is changed after track creation 5559 if (mActiveTrack != 0) { 5560 status = INVALID_OPERATION; 5561 } else { 5562 reconfig = true; 5563 } 5564 } 5565 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5566 // forward device change to effects that have requested to be 5567 // aware of attached audio device. 5568 for (size_t i = 0; i < mEffectChains.size(); i++) { 5569 mEffectChains[i]->setDevice_l(value); 5570 } 5571 // store input device and output device but do not forward output device to audio HAL. 5572 // Note that status is ignored by the caller for output device 5573 // (see AudioFlinger::setParameters() 5574 if (value & AUDIO_DEVICE_OUT_ALL) { 5575 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5576 status = BAD_VALUE; 5577 } else { 5578 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5579 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5580 if (mTrack != NULL) { 5581 bool suspend = audio_is_bluetooth_sco_device( 5582 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5583 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5584 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5585 } 5586 } 5587 mDevice |= (uint32_t)value; 5588 } 5589 if (status == NO_ERROR) { 5590 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5591 if (status == INVALID_OPERATION) { 5592 mInput->stream->common.standby(&mInput->stream->common); 5593 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5594 keyValuePair.string()); 5595 } 5596 if (reconfig) { 5597 if (status == BAD_VALUE && 5598 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5599 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5600 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5601 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5602 (reqChannelCount <= FCC_2)) { 5603 status = NO_ERROR; 5604 } 5605 if (status == NO_ERROR) { 5606 readInputParameters(); 5607 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5608 } 5609 } 5610 } 5611 5612 mNewParameters.removeAt(0); 5613 5614 mParamStatus = status; 5615 mParamCond.signal(); 5616 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5617 // already timed out waiting for the status and will never signal the condition. 5618 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5619 } 5620 return reconfig; 5621} 5622 5623String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5624{ 5625 char *s; 5626 String8 out_s8 = String8(); 5627 5628 Mutex::Autolock _l(mLock); 5629 if (initCheck() != NO_ERROR) { 5630 return out_s8; 5631 } 5632 5633 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5634 out_s8 = String8(s); 5635 free(s); 5636 return out_s8; 5637} 5638 5639void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5640 AudioSystem::OutputDescriptor desc; 5641 void *param2 = NULL; 5642 5643 switch (event) { 5644 case AudioSystem::INPUT_OPENED: 5645 case AudioSystem::INPUT_CONFIG_CHANGED: 5646 desc.channels = mChannelMask; 5647 desc.samplingRate = mSampleRate; 5648 desc.format = mFormat; 5649 desc.frameCount = mFrameCount; 5650 desc.latency = 0; 5651 param2 = &desc; 5652 break; 5653 5654 case AudioSystem::INPUT_CLOSED: 5655 default: 5656 break; 5657 } 5658 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5659} 5660 5661void AudioFlinger::RecordThread::readInputParameters() 5662{ 5663 delete mRsmpInBuffer; 5664 // mRsmpInBuffer is always assigned a new[] below 5665 delete mRsmpOutBuffer; 5666 mRsmpOutBuffer = NULL; 5667 delete mResampler; 5668 mResampler = NULL; 5669 5670 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5671 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5672 mChannelCount = (uint16_t)popcount(mChannelMask); 5673 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5674 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5675 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5676 mFrameCount = mInputBytes / mFrameSize; 5677 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5678 5679 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5680 { 5681 int channelCount; 5682 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5683 // stereo to mono post process as the resampler always outputs stereo. 5684 if (mChannelCount == 1 && mReqChannelCount == 2) { 5685 channelCount = 1; 5686 } else { 5687 channelCount = 2; 5688 } 5689 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5690 mResampler->setSampleRate(mSampleRate); 5691 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5692 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5693 5694 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5695 if (mChannelCount == 1 && mReqChannelCount == 1) { 5696 mFrameCount >>= 1; 5697 } 5698 5699 } 5700 mRsmpInIndex = mFrameCount; 5701} 5702 5703unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5704{ 5705 Mutex::Autolock _l(mLock); 5706 if (initCheck() != NO_ERROR) { 5707 return 0; 5708 } 5709 5710 return mInput->stream->get_input_frames_lost(mInput->stream); 5711} 5712 5713uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5714{ 5715 Mutex::Autolock _l(mLock); 5716 uint32_t result = 0; 5717 if (getEffectChain_l(sessionId) != 0) { 5718 result = EFFECT_SESSION; 5719 } 5720 5721 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5722 result |= TRACK_SESSION; 5723 } 5724 5725 return result; 5726} 5727 5728AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5729{ 5730 Mutex::Autolock _l(mLock); 5731 return mTrack; 5732} 5733 5734AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5735{ 5736 Mutex::Autolock _l(mLock); 5737 return mInput; 5738} 5739 5740AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5741{ 5742 Mutex::Autolock _l(mLock); 5743 AudioStreamIn *input = mInput; 5744 mInput = NULL; 5745 return input; 5746} 5747 5748// this method must always be called either with ThreadBase mLock held or inside the thread loop 5749audio_stream_t* AudioFlinger::RecordThread::stream() const 5750{ 5751 if (mInput == NULL) { 5752 return NULL; 5753 } 5754 return &mInput->stream->common; 5755} 5756 5757 5758// ---------------------------------------------------------------------------- 5759 5760audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 5761{ 5762 if (!settingsAllowed()) { 5763 return 0; 5764 } 5765 Mutex::Autolock _l(mLock); 5766 return loadHwModule_l(name); 5767} 5768 5769// loadHwModule_l() must be called with AudioFlinger::mLock held 5770audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 5771{ 5772 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5773 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 5774 ALOGW("loadHwModule() module %s already loaded", name); 5775 return mAudioHwDevs.keyAt(i); 5776 } 5777 } 5778 5779 audio_hw_device_t *dev; 5780 5781 int rc = load_audio_interface(name, &dev); 5782 if (rc) { 5783 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 5784 return 0; 5785 } 5786 5787 mHardwareStatus = AUDIO_HW_INIT; 5788 rc = dev->init_check(dev); 5789 mHardwareStatus = AUDIO_HW_IDLE; 5790 if (rc) { 5791 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 5792 return 0; 5793 } 5794 5795 if ((mMasterVolumeSupportLvl != MVS_NONE) && 5796 (NULL != dev->set_master_volume)) { 5797 AutoMutex lock(mHardwareLock); 5798 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5799 dev->set_master_volume(dev, mMasterVolume); 5800 mHardwareStatus = AUDIO_HW_IDLE; 5801 } 5802 5803 audio_module_handle_t handle = nextUniqueId(); 5804 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 5805 5806 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 5807 name, dev->common.module->name, dev->common.module->id, handle); 5808 5809 return handle; 5810 5811} 5812 5813audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 5814 audio_devices_t *pDevices, 5815 uint32_t *pSamplingRate, 5816 audio_format_t *pFormat, 5817 audio_channel_mask_t *pChannelMask, 5818 uint32_t *pLatencyMs, 5819 audio_output_flags_t flags) 5820{ 5821 status_t status; 5822 PlaybackThread *thread = NULL; 5823 struct audio_config config = { 5824 sample_rate: pSamplingRate ? *pSamplingRate : 0, 5825 channel_mask: pChannelMask ? *pChannelMask : 0, 5826 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 5827 }; 5828 audio_stream_out_t *outStream = NULL; 5829 audio_hw_device_t *outHwDev; 5830 5831 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5832 module, 5833 (pDevices != NULL) ? (int)*pDevices : 0, 5834 config.sample_rate, 5835 config.format, 5836 config.channel_mask, 5837 flags); 5838 5839 if (pDevices == NULL || *pDevices == 0) { 5840 return 0; 5841 } 5842 5843 Mutex::Autolock _l(mLock); 5844 5845 outHwDev = findSuitableHwDev_l(module, *pDevices); 5846 if (outHwDev == NULL) 5847 return 0; 5848 5849 audio_io_handle_t id = nextUniqueId(); 5850 5851 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5852 5853 status = outHwDev->open_output_stream(outHwDev, 5854 id, 5855 *pDevices, 5856 (audio_output_flags_t)flags, 5857 &config, 5858 &outStream); 5859 5860 mHardwareStatus = AUDIO_HW_IDLE; 5861 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5862 outStream, 5863 config.sample_rate, 5864 config.format, 5865 config.channel_mask, 5866 status); 5867 5868 if (status == NO_ERROR && outStream != NULL) { 5869 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5870 5871 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 5872 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 5873 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 5874 thread = new DirectOutputThread(this, output, id, *pDevices); 5875 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5876 } else { 5877 thread = new MixerThread(this, output, id, *pDevices); 5878 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5879 } 5880 mPlaybackThreads.add(id, thread); 5881 5882 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 5883 if (pFormat != NULL) *pFormat = config.format; 5884 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 5885 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5886 5887 // notify client processes of the new output creation 5888 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5889 5890 // the first primary output opened designates the primary hw device 5891 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 5892 ALOGI("Using module %d has the primary audio interface", module); 5893 mPrimaryHardwareDev = outHwDev; 5894 5895 AutoMutex lock(mHardwareLock); 5896 mHardwareStatus = AUDIO_HW_SET_MODE; 5897 outHwDev->set_mode(outHwDev, mMode); 5898 5899 // Determine the level of master volume support the primary audio HAL has, 5900 // and set the initial master volume at the same time. 5901 float initialVolume = 1.0; 5902 mMasterVolumeSupportLvl = MVS_NONE; 5903 5904 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 5905 if ((NULL != outHwDev->get_master_volume) && 5906 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 5907 mMasterVolumeSupportLvl = MVS_FULL; 5908 } else { 5909 mMasterVolumeSupportLvl = MVS_SETONLY; 5910 initialVolume = 1.0; 5911 } 5912 5913 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5914 if ((NULL == outHwDev->set_master_volume) || 5915 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 5916 mMasterVolumeSupportLvl = MVS_NONE; 5917 } 5918 // now that we have a primary device, initialize master volume on other devices 5919 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5920 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 5921 5922 if ((dev != mPrimaryHardwareDev) && 5923 (NULL != dev->set_master_volume)) { 5924 dev->set_master_volume(dev, initialVolume); 5925 } 5926 } 5927 mHardwareStatus = AUDIO_HW_IDLE; 5928 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 5929 ? initialVolume 5930 : 1.0; 5931 mMasterVolume = initialVolume; 5932 } 5933 return id; 5934 } 5935 5936 return 0; 5937} 5938 5939audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5940 audio_io_handle_t output2) 5941{ 5942 Mutex::Autolock _l(mLock); 5943 MixerThread *thread1 = checkMixerThread_l(output1); 5944 MixerThread *thread2 = checkMixerThread_l(output2); 5945 5946 if (thread1 == NULL || thread2 == NULL) { 5947 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5948 return 0; 5949 } 5950 5951 audio_io_handle_t id = nextUniqueId(); 5952 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5953 thread->addOutputTrack(thread2); 5954 mPlaybackThreads.add(id, thread); 5955 // notify client processes of the new output creation 5956 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5957 return id; 5958} 5959 5960status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5961{ 5962 // keep strong reference on the playback thread so that 5963 // it is not destroyed while exit() is executed 5964 sp<PlaybackThread> thread; 5965 { 5966 Mutex::Autolock _l(mLock); 5967 thread = checkPlaybackThread_l(output); 5968 if (thread == NULL) { 5969 return BAD_VALUE; 5970 } 5971 5972 ALOGV("closeOutput() %d", output); 5973 5974 if (thread->type() == ThreadBase::MIXER) { 5975 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5976 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5977 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5978 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5979 } 5980 } 5981 } 5982 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5983 mPlaybackThreads.removeItem(output); 5984 } 5985 thread->exit(); 5986 // The thread entity (active unit of execution) is no longer running here, 5987 // but the ThreadBase container still exists. 5988 5989 if (thread->type() != ThreadBase::DUPLICATING) { 5990 AudioStreamOut *out = thread->clearOutput(); 5991 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5992 // from now on thread->mOutput is NULL 5993 out->hwDev->close_output_stream(out->hwDev, out->stream); 5994 delete out; 5995 } 5996 return NO_ERROR; 5997} 5998 5999status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6000{ 6001 Mutex::Autolock _l(mLock); 6002 PlaybackThread *thread = checkPlaybackThread_l(output); 6003 6004 if (thread == NULL) { 6005 return BAD_VALUE; 6006 } 6007 6008 ALOGV("suspendOutput() %d", output); 6009 thread->suspend(); 6010 6011 return NO_ERROR; 6012} 6013 6014status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6015{ 6016 Mutex::Autolock _l(mLock); 6017 PlaybackThread *thread = checkPlaybackThread_l(output); 6018 6019 if (thread == NULL) { 6020 return BAD_VALUE; 6021 } 6022 6023 ALOGV("restoreOutput() %d", output); 6024 6025 thread->restore(); 6026 6027 return NO_ERROR; 6028} 6029 6030audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6031 audio_devices_t *pDevices, 6032 uint32_t *pSamplingRate, 6033 audio_format_t *pFormat, 6034 uint32_t *pChannelMask) 6035{ 6036 status_t status; 6037 RecordThread *thread = NULL; 6038 struct audio_config config = { 6039 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6040 channel_mask: pChannelMask ? *pChannelMask : 0, 6041 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6042 }; 6043 uint32_t reqSamplingRate = config.sample_rate; 6044 audio_format_t reqFormat = config.format; 6045 audio_channel_mask_t reqChannels = config.channel_mask; 6046 audio_stream_in_t *inStream = NULL; 6047 audio_hw_device_t *inHwDev; 6048 6049 if (pDevices == NULL || *pDevices == 0) { 6050 return 0; 6051 } 6052 6053 Mutex::Autolock _l(mLock); 6054 6055 inHwDev = findSuitableHwDev_l(module, *pDevices); 6056 if (inHwDev == NULL) 6057 return 0; 6058 6059 audio_io_handle_t id = nextUniqueId(); 6060 6061 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6062 &inStream); 6063 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6064 inStream, 6065 config.sample_rate, 6066 config.format, 6067 config.channel_mask, 6068 status); 6069 6070 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6071 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6072 // or stereo to mono conversions on 16 bit PCM inputs. 6073 if (status == BAD_VALUE && 6074 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6075 (config.sample_rate <= 2 * reqSamplingRate) && 6076 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6077 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6078 inStream = NULL; 6079 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6080 } 6081 6082 if (status == NO_ERROR && inStream != NULL) { 6083 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6084 6085 // Start record thread 6086 // RecorThread require both input and output device indication to forward to audio 6087 // pre processing modules 6088 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6089 thread = new RecordThread(this, 6090 input, 6091 reqSamplingRate, 6092 reqChannels, 6093 id, 6094 device); 6095 mRecordThreads.add(id, thread); 6096 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6097 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6098 if (pFormat != NULL) *pFormat = config.format; 6099 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6100 6101 input->stream->common.standby(&input->stream->common); 6102 6103 // notify client processes of the new input creation 6104 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6105 return id; 6106 } 6107 6108 return 0; 6109} 6110 6111status_t AudioFlinger::closeInput(audio_io_handle_t input) 6112{ 6113 // keep strong reference on the record thread so that 6114 // it is not destroyed while exit() is executed 6115 sp<RecordThread> thread; 6116 { 6117 Mutex::Autolock _l(mLock); 6118 thread = checkRecordThread_l(input); 6119 if (thread == NULL) { 6120 return BAD_VALUE; 6121 } 6122 6123 ALOGV("closeInput() %d", input); 6124 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6125 mRecordThreads.removeItem(input); 6126 } 6127 thread->exit(); 6128 // The thread entity (active unit of execution) is no longer running here, 6129 // but the ThreadBase container still exists. 6130 6131 AudioStreamIn *in = thread->clearInput(); 6132 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6133 // from now on thread->mInput is NULL 6134 in->hwDev->close_input_stream(in->hwDev, in->stream); 6135 delete in; 6136 6137 return NO_ERROR; 6138} 6139 6140status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6141{ 6142 Mutex::Autolock _l(mLock); 6143 MixerThread *dstThread = checkMixerThread_l(output); 6144 if (dstThread == NULL) { 6145 ALOGW("setStreamOutput() bad output id %d", output); 6146 return BAD_VALUE; 6147 } 6148 6149 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6150 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6151 6152 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6153 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6154 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6155 MixerThread *srcThread = (MixerThread *)thread; 6156 srcThread->invalidateTracks(stream); 6157 } 6158 } 6159 6160 return NO_ERROR; 6161} 6162 6163 6164int AudioFlinger::newAudioSessionId() 6165{ 6166 return nextUniqueId(); 6167} 6168 6169void AudioFlinger::acquireAudioSessionId(int audioSession) 6170{ 6171 Mutex::Autolock _l(mLock); 6172 pid_t caller = IPCThreadState::self()->getCallingPid(); 6173 ALOGV("acquiring %d from %d", audioSession, caller); 6174 size_t num = mAudioSessionRefs.size(); 6175 for (size_t i = 0; i< num; i++) { 6176 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6177 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6178 ref->mCnt++; 6179 ALOGV(" incremented refcount to %d", ref->mCnt); 6180 return; 6181 } 6182 } 6183 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6184 ALOGV(" added new entry for %d", audioSession); 6185} 6186 6187void AudioFlinger::releaseAudioSessionId(int audioSession) 6188{ 6189 Mutex::Autolock _l(mLock); 6190 pid_t caller = IPCThreadState::self()->getCallingPid(); 6191 ALOGV("releasing %d from %d", audioSession, caller); 6192 size_t num = mAudioSessionRefs.size(); 6193 for (size_t i = 0; i< num; i++) { 6194 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6195 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6196 ref->mCnt--; 6197 ALOGV(" decremented refcount to %d", ref->mCnt); 6198 if (ref->mCnt == 0) { 6199 mAudioSessionRefs.removeAt(i); 6200 delete ref; 6201 purgeStaleEffects_l(); 6202 } 6203 return; 6204 } 6205 } 6206 ALOGW("session id %d not found for pid %d", audioSession, caller); 6207} 6208 6209void AudioFlinger::purgeStaleEffects_l() { 6210 6211 ALOGV("purging stale effects"); 6212 6213 Vector< sp<EffectChain> > chains; 6214 6215 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6216 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6217 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6218 sp<EffectChain> ec = t->mEffectChains[j]; 6219 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6220 chains.push(ec); 6221 } 6222 } 6223 } 6224 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6225 sp<RecordThread> t = mRecordThreads.valueAt(i); 6226 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6227 sp<EffectChain> ec = t->mEffectChains[j]; 6228 chains.push(ec); 6229 } 6230 } 6231 6232 for (size_t i = 0; i < chains.size(); i++) { 6233 sp<EffectChain> ec = chains[i]; 6234 int sessionid = ec->sessionId(); 6235 sp<ThreadBase> t = ec->mThread.promote(); 6236 if (t == 0) { 6237 continue; 6238 } 6239 size_t numsessionrefs = mAudioSessionRefs.size(); 6240 bool found = false; 6241 for (size_t k = 0; k < numsessionrefs; k++) { 6242 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6243 if (ref->mSessionid == sessionid) { 6244 ALOGV(" session %d still exists for %d with %d refs", 6245 sessionid, ref->mPid, ref->mCnt); 6246 found = true; 6247 break; 6248 } 6249 } 6250 if (!found) { 6251 // remove all effects from the chain 6252 while (ec->mEffects.size()) { 6253 sp<EffectModule> effect = ec->mEffects[0]; 6254 effect->unPin(); 6255 Mutex::Autolock _l (t->mLock); 6256 t->removeEffect_l(effect); 6257 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6258 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6259 if (handle != 0) { 6260 handle->mEffect.clear(); 6261 if (handle->mHasControl && handle->mEnabled) { 6262 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6263 } 6264 } 6265 } 6266 AudioSystem::unregisterEffect(effect->id()); 6267 } 6268 } 6269 } 6270 return; 6271} 6272 6273// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6274AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6275{ 6276 return mPlaybackThreads.valueFor(output).get(); 6277} 6278 6279// checkMixerThread_l() must be called with AudioFlinger::mLock held 6280AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6281{ 6282 PlaybackThread *thread = checkPlaybackThread_l(output); 6283 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6284} 6285 6286// checkRecordThread_l() must be called with AudioFlinger::mLock held 6287AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6288{ 6289 return mRecordThreads.valueFor(input).get(); 6290} 6291 6292uint32_t AudioFlinger::nextUniqueId() 6293{ 6294 return android_atomic_inc(&mNextUniqueId); 6295} 6296 6297AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6298{ 6299 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6300 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6301 AudioStreamOut *output = thread->getOutput(); 6302 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6303 return thread; 6304 } 6305 } 6306 return NULL; 6307} 6308 6309uint32_t AudioFlinger::primaryOutputDevice_l() const 6310{ 6311 PlaybackThread *thread = primaryPlaybackThread_l(); 6312 6313 if (thread == NULL) { 6314 return 0; 6315 } 6316 6317 return thread->device(); 6318} 6319 6320sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6321 int triggerSession, 6322 int listenerSession, 6323 sync_event_callback_t callBack, 6324 void *cookie) 6325{ 6326 Mutex::Autolock _l(mLock); 6327 6328 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6329 status_t playStatus = NAME_NOT_FOUND; 6330 status_t recStatus = NAME_NOT_FOUND; 6331 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6332 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6333 if (playStatus == NO_ERROR) { 6334 return event; 6335 } 6336 } 6337 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6338 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6339 if (recStatus == NO_ERROR) { 6340 return event; 6341 } 6342 } 6343 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6344 mPendingSyncEvents.add(event); 6345 } else { 6346 ALOGV("createSyncEvent() invalid event %d", event->type()); 6347 event.clear(); 6348 } 6349 return event; 6350} 6351 6352// ---------------------------------------------------------------------------- 6353// Effect management 6354// ---------------------------------------------------------------------------- 6355 6356 6357status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6358{ 6359 Mutex::Autolock _l(mLock); 6360 return EffectQueryNumberEffects(numEffects); 6361} 6362 6363status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6364{ 6365 Mutex::Autolock _l(mLock); 6366 return EffectQueryEffect(index, descriptor); 6367} 6368 6369status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6370 effect_descriptor_t *descriptor) const 6371{ 6372 Mutex::Autolock _l(mLock); 6373 return EffectGetDescriptor(pUuid, descriptor); 6374} 6375 6376 6377sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6378 effect_descriptor_t *pDesc, 6379 const sp<IEffectClient>& effectClient, 6380 int32_t priority, 6381 audio_io_handle_t io, 6382 int sessionId, 6383 status_t *status, 6384 int *id, 6385 int *enabled) 6386{ 6387 status_t lStatus = NO_ERROR; 6388 sp<EffectHandle> handle; 6389 effect_descriptor_t desc; 6390 6391 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6392 pid, effectClient.get(), priority, sessionId, io); 6393 6394 if (pDesc == NULL) { 6395 lStatus = BAD_VALUE; 6396 goto Exit; 6397 } 6398 6399 // check audio settings permission for global effects 6400 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6401 lStatus = PERMISSION_DENIED; 6402 goto Exit; 6403 } 6404 6405 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6406 // that can only be created by audio policy manager (running in same process) 6407 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6408 lStatus = PERMISSION_DENIED; 6409 goto Exit; 6410 } 6411 6412 if (io == 0) { 6413 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6414 // output must be specified by AudioPolicyManager when using session 6415 // AUDIO_SESSION_OUTPUT_STAGE 6416 lStatus = BAD_VALUE; 6417 goto Exit; 6418 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6419 // if the output returned by getOutputForEffect() is removed before we lock the 6420 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6421 // and we will exit safely 6422 io = AudioSystem::getOutputForEffect(&desc); 6423 } 6424 } 6425 6426 { 6427 Mutex::Autolock _l(mLock); 6428 6429 6430 if (!EffectIsNullUuid(&pDesc->uuid)) { 6431 // if uuid is specified, request effect descriptor 6432 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6433 if (lStatus < 0) { 6434 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6435 goto Exit; 6436 } 6437 } else { 6438 // if uuid is not specified, look for an available implementation 6439 // of the required type in effect factory 6440 if (EffectIsNullUuid(&pDesc->type)) { 6441 ALOGW("createEffect() no effect type"); 6442 lStatus = BAD_VALUE; 6443 goto Exit; 6444 } 6445 uint32_t numEffects = 0; 6446 effect_descriptor_t d; 6447 d.flags = 0; // prevent compiler warning 6448 bool found = false; 6449 6450 lStatus = EffectQueryNumberEffects(&numEffects); 6451 if (lStatus < 0) { 6452 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6453 goto Exit; 6454 } 6455 for (uint32_t i = 0; i < numEffects; i++) { 6456 lStatus = EffectQueryEffect(i, &desc); 6457 if (lStatus < 0) { 6458 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6459 continue; 6460 } 6461 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6462 // If matching type found save effect descriptor. If the session is 6463 // 0 and the effect is not auxiliary, continue enumeration in case 6464 // an auxiliary version of this effect type is available 6465 found = true; 6466 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6467 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6468 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6469 break; 6470 } 6471 } 6472 } 6473 if (!found) { 6474 lStatus = BAD_VALUE; 6475 ALOGW("createEffect() effect not found"); 6476 goto Exit; 6477 } 6478 // For same effect type, chose auxiliary version over insert version if 6479 // connect to output mix (Compliance to OpenSL ES) 6480 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6481 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6482 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6483 } 6484 } 6485 6486 // Do not allow auxiliary effects on a session different from 0 (output mix) 6487 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6488 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6489 lStatus = INVALID_OPERATION; 6490 goto Exit; 6491 } 6492 6493 // check recording permission for visualizer 6494 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6495 !recordingAllowed()) { 6496 lStatus = PERMISSION_DENIED; 6497 goto Exit; 6498 } 6499 6500 // return effect descriptor 6501 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6502 6503 // If output is not specified try to find a matching audio session ID in one of the 6504 // output threads. 6505 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6506 // because of code checking output when entering the function. 6507 // Note: io is never 0 when creating an effect on an input 6508 if (io == 0) { 6509 // look for the thread where the specified audio session is present 6510 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6511 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6512 io = mPlaybackThreads.keyAt(i); 6513 break; 6514 } 6515 } 6516 if (io == 0) { 6517 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6518 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6519 io = mRecordThreads.keyAt(i); 6520 break; 6521 } 6522 } 6523 } 6524 // If no output thread contains the requested session ID, default to 6525 // first output. The effect chain will be moved to the correct output 6526 // thread when a track with the same session ID is created 6527 if (io == 0 && mPlaybackThreads.size()) { 6528 io = mPlaybackThreads.keyAt(0); 6529 } 6530 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6531 } 6532 ThreadBase *thread = checkRecordThread_l(io); 6533 if (thread == NULL) { 6534 thread = checkPlaybackThread_l(io); 6535 if (thread == NULL) { 6536 ALOGE("createEffect() unknown output thread"); 6537 lStatus = BAD_VALUE; 6538 goto Exit; 6539 } 6540 } 6541 6542 sp<Client> client = registerPid_l(pid); 6543 6544 // create effect on selected output thread 6545 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6546 &desc, enabled, &lStatus); 6547 if (handle != 0 && id != NULL) { 6548 *id = handle->id(); 6549 } 6550 } 6551 6552Exit: 6553 if (status != NULL) { 6554 *status = lStatus; 6555 } 6556 return handle; 6557} 6558 6559status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6560 audio_io_handle_t dstOutput) 6561{ 6562 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6563 sessionId, srcOutput, dstOutput); 6564 Mutex::Autolock _l(mLock); 6565 if (srcOutput == dstOutput) { 6566 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6567 return NO_ERROR; 6568 } 6569 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6570 if (srcThread == NULL) { 6571 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6572 return BAD_VALUE; 6573 } 6574 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6575 if (dstThread == NULL) { 6576 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6577 return BAD_VALUE; 6578 } 6579 6580 Mutex::Autolock _dl(dstThread->mLock); 6581 Mutex::Autolock _sl(srcThread->mLock); 6582 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6583 6584 return NO_ERROR; 6585} 6586 6587// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6588status_t AudioFlinger::moveEffectChain_l(int sessionId, 6589 AudioFlinger::PlaybackThread *srcThread, 6590 AudioFlinger::PlaybackThread *dstThread, 6591 bool reRegister) 6592{ 6593 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6594 sessionId, srcThread, dstThread); 6595 6596 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6597 if (chain == 0) { 6598 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6599 sessionId, srcThread); 6600 return INVALID_OPERATION; 6601 } 6602 6603 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6604 // so that a new chain is created with correct parameters when first effect is added. This is 6605 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6606 // removed. 6607 srcThread->removeEffectChain_l(chain); 6608 6609 // transfer all effects one by one so that new effect chain is created on new thread with 6610 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6611 audio_io_handle_t dstOutput = dstThread->id(); 6612 sp<EffectChain> dstChain; 6613 uint32_t strategy = 0; // prevent compiler warning 6614 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6615 while (effect != 0) { 6616 srcThread->removeEffect_l(effect); 6617 dstThread->addEffect_l(effect); 6618 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6619 if (effect->state() == EffectModule::ACTIVE || 6620 effect->state() == EffectModule::STOPPING) { 6621 effect->start(); 6622 } 6623 // if the move request is not received from audio policy manager, the effect must be 6624 // re-registered with the new strategy and output 6625 if (dstChain == 0) { 6626 dstChain = effect->chain().promote(); 6627 if (dstChain == 0) { 6628 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6629 srcThread->addEffect_l(effect); 6630 return NO_INIT; 6631 } 6632 strategy = dstChain->strategy(); 6633 } 6634 if (reRegister) { 6635 AudioSystem::unregisterEffect(effect->id()); 6636 AudioSystem::registerEffect(&effect->desc(), 6637 dstOutput, 6638 strategy, 6639 sessionId, 6640 effect->id()); 6641 } 6642 effect = chain->getEffectFromId_l(0); 6643 } 6644 6645 return NO_ERROR; 6646} 6647 6648 6649// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6650sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6651 const sp<AudioFlinger::Client>& client, 6652 const sp<IEffectClient>& effectClient, 6653 int32_t priority, 6654 int sessionId, 6655 effect_descriptor_t *desc, 6656 int *enabled, 6657 status_t *status 6658 ) 6659{ 6660 sp<EffectModule> effect; 6661 sp<EffectHandle> handle; 6662 status_t lStatus; 6663 sp<EffectChain> chain; 6664 bool chainCreated = false; 6665 bool effectCreated = false; 6666 bool effectRegistered = false; 6667 6668 lStatus = initCheck(); 6669 if (lStatus != NO_ERROR) { 6670 ALOGW("createEffect_l() Audio driver not initialized."); 6671 goto Exit; 6672 } 6673 6674 // Do not allow effects with session ID 0 on direct output or duplicating threads 6675 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6676 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6677 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6678 desc->name, sessionId); 6679 lStatus = BAD_VALUE; 6680 goto Exit; 6681 } 6682 // Only Pre processor effects are allowed on input threads and only on input threads 6683 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6684 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6685 desc->name, desc->flags, mType); 6686 lStatus = BAD_VALUE; 6687 goto Exit; 6688 } 6689 6690 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6691 6692 { // scope for mLock 6693 Mutex::Autolock _l(mLock); 6694 6695 // check for existing effect chain with the requested audio session 6696 chain = getEffectChain_l(sessionId); 6697 if (chain == 0) { 6698 // create a new chain for this session 6699 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6700 chain = new EffectChain(this, sessionId); 6701 addEffectChain_l(chain); 6702 chain->setStrategy(getStrategyForSession_l(sessionId)); 6703 chainCreated = true; 6704 } else { 6705 effect = chain->getEffectFromDesc_l(desc); 6706 } 6707 6708 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6709 6710 if (effect == 0) { 6711 int id = mAudioFlinger->nextUniqueId(); 6712 // Check CPU and memory usage 6713 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6714 if (lStatus != NO_ERROR) { 6715 goto Exit; 6716 } 6717 effectRegistered = true; 6718 // create a new effect module if none present in the chain 6719 effect = new EffectModule(this, chain, desc, id, sessionId); 6720 lStatus = effect->status(); 6721 if (lStatus != NO_ERROR) { 6722 goto Exit; 6723 } 6724 lStatus = chain->addEffect_l(effect); 6725 if (lStatus != NO_ERROR) { 6726 goto Exit; 6727 } 6728 effectCreated = true; 6729 6730 effect->setDevice(mDevice); 6731 effect->setMode(mAudioFlinger->getMode()); 6732 } 6733 // create effect handle and connect it to effect module 6734 handle = new EffectHandle(effect, client, effectClient, priority); 6735 lStatus = effect->addHandle(handle); 6736 if (enabled != NULL) { 6737 *enabled = (int)effect->isEnabled(); 6738 } 6739 } 6740 6741Exit: 6742 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6743 Mutex::Autolock _l(mLock); 6744 if (effectCreated) { 6745 chain->removeEffect_l(effect); 6746 } 6747 if (effectRegistered) { 6748 AudioSystem::unregisterEffect(effect->id()); 6749 } 6750 if (chainCreated) { 6751 removeEffectChain_l(chain); 6752 } 6753 handle.clear(); 6754 } 6755 6756 if (status != NULL) { 6757 *status = lStatus; 6758 } 6759 return handle; 6760} 6761 6762sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6763{ 6764 sp<EffectChain> chain = getEffectChain_l(sessionId); 6765 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6766} 6767 6768// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6769// PlaybackThread::mLock held 6770status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6771{ 6772 // check for existing effect chain with the requested audio session 6773 int sessionId = effect->sessionId(); 6774 sp<EffectChain> chain = getEffectChain_l(sessionId); 6775 bool chainCreated = false; 6776 6777 if (chain == 0) { 6778 // create a new chain for this session 6779 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6780 chain = new EffectChain(this, sessionId); 6781 addEffectChain_l(chain); 6782 chain->setStrategy(getStrategyForSession_l(sessionId)); 6783 chainCreated = true; 6784 } 6785 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6786 6787 if (chain->getEffectFromId_l(effect->id()) != 0) { 6788 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6789 this, effect->desc().name, chain.get()); 6790 return BAD_VALUE; 6791 } 6792 6793 status_t status = chain->addEffect_l(effect); 6794 if (status != NO_ERROR) { 6795 if (chainCreated) { 6796 removeEffectChain_l(chain); 6797 } 6798 return status; 6799 } 6800 6801 effect->setDevice(mDevice); 6802 effect->setMode(mAudioFlinger->getMode()); 6803 return NO_ERROR; 6804} 6805 6806void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6807 6808 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6809 effect_descriptor_t desc = effect->desc(); 6810 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6811 detachAuxEffect_l(effect->id()); 6812 } 6813 6814 sp<EffectChain> chain = effect->chain().promote(); 6815 if (chain != 0) { 6816 // remove effect chain if removing last effect 6817 if (chain->removeEffect_l(effect) == 0) { 6818 removeEffectChain_l(chain); 6819 } 6820 } else { 6821 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6822 } 6823} 6824 6825void AudioFlinger::ThreadBase::lockEffectChains_l( 6826 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6827{ 6828 effectChains = mEffectChains; 6829 for (size_t i = 0; i < mEffectChains.size(); i++) { 6830 mEffectChains[i]->lock(); 6831 } 6832} 6833 6834void AudioFlinger::ThreadBase::unlockEffectChains( 6835 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6836{ 6837 for (size_t i = 0; i < effectChains.size(); i++) { 6838 effectChains[i]->unlock(); 6839 } 6840} 6841 6842sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6843{ 6844 Mutex::Autolock _l(mLock); 6845 return getEffectChain_l(sessionId); 6846} 6847 6848sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6849{ 6850 size_t size = mEffectChains.size(); 6851 for (size_t i = 0; i < size; i++) { 6852 if (mEffectChains[i]->sessionId() == sessionId) { 6853 return mEffectChains[i]; 6854 } 6855 } 6856 return 0; 6857} 6858 6859void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6860{ 6861 Mutex::Autolock _l(mLock); 6862 size_t size = mEffectChains.size(); 6863 for (size_t i = 0; i < size; i++) { 6864 mEffectChains[i]->setMode_l(mode); 6865 } 6866} 6867 6868void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6869 const wp<EffectHandle>& handle, 6870 bool unpinIfLast) { 6871 6872 Mutex::Autolock _l(mLock); 6873 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6874 // delete the effect module if removing last handle on it 6875 if (effect->removeHandle(handle) == 0) { 6876 if (!effect->isPinned() || unpinIfLast) { 6877 removeEffect_l(effect); 6878 AudioSystem::unregisterEffect(effect->id()); 6879 } 6880 } 6881} 6882 6883status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6884{ 6885 int session = chain->sessionId(); 6886 int16_t *buffer = mMixBuffer; 6887 bool ownsBuffer = false; 6888 6889 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6890 if (session > 0) { 6891 // Only one effect chain can be present in direct output thread and it uses 6892 // the mix buffer as input 6893 if (mType != DIRECT) { 6894 size_t numSamples = mFrameCount * mChannelCount; 6895 buffer = new int16_t[numSamples]; 6896 memset(buffer, 0, numSamples * sizeof(int16_t)); 6897 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6898 ownsBuffer = true; 6899 } 6900 6901 // Attach all tracks with same session ID to this chain. 6902 for (size_t i = 0; i < mTracks.size(); ++i) { 6903 sp<Track> track = mTracks[i]; 6904 if (session == track->sessionId()) { 6905 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6906 track->setMainBuffer(buffer); 6907 chain->incTrackCnt(); 6908 } 6909 } 6910 6911 // indicate all active tracks in the chain 6912 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6913 sp<Track> track = mActiveTracks[i].promote(); 6914 if (track == 0) continue; 6915 if (session == track->sessionId()) { 6916 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6917 chain->incActiveTrackCnt(); 6918 } 6919 } 6920 } 6921 6922 chain->setInBuffer(buffer, ownsBuffer); 6923 chain->setOutBuffer(mMixBuffer); 6924 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6925 // chains list in order to be processed last as it contains output stage effects 6926 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6927 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6928 // after track specific effects and before output stage 6929 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6930 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6931 // Effect chain for other sessions are inserted at beginning of effect 6932 // chains list to be processed before output mix effects. Relative order between other 6933 // sessions is not important 6934 size_t size = mEffectChains.size(); 6935 size_t i = 0; 6936 for (i = 0; i < size; i++) { 6937 if (mEffectChains[i]->sessionId() < session) break; 6938 } 6939 mEffectChains.insertAt(chain, i); 6940 checkSuspendOnAddEffectChain_l(chain); 6941 6942 return NO_ERROR; 6943} 6944 6945size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6946{ 6947 int session = chain->sessionId(); 6948 6949 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6950 6951 for (size_t i = 0; i < mEffectChains.size(); i++) { 6952 if (chain == mEffectChains[i]) { 6953 mEffectChains.removeAt(i); 6954 // detach all active tracks from the chain 6955 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6956 sp<Track> track = mActiveTracks[i].promote(); 6957 if (track == 0) continue; 6958 if (session == track->sessionId()) { 6959 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6960 chain.get(), session); 6961 chain->decActiveTrackCnt(); 6962 } 6963 } 6964 6965 // detach all tracks with same session ID from this chain 6966 for (size_t i = 0; i < mTracks.size(); ++i) { 6967 sp<Track> track = mTracks[i]; 6968 if (session == track->sessionId()) { 6969 track->setMainBuffer(mMixBuffer); 6970 chain->decTrackCnt(); 6971 } 6972 } 6973 break; 6974 } 6975 } 6976 return mEffectChains.size(); 6977} 6978 6979status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6980 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6981{ 6982 Mutex::Autolock _l(mLock); 6983 return attachAuxEffect_l(track, EffectId); 6984} 6985 6986status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6987 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6988{ 6989 status_t status = NO_ERROR; 6990 6991 if (EffectId == 0) { 6992 track->setAuxBuffer(0, NULL); 6993 } else { 6994 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6995 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6996 if (effect != 0) { 6997 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6998 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6999 } else { 7000 status = INVALID_OPERATION; 7001 } 7002 } else { 7003 status = BAD_VALUE; 7004 } 7005 } 7006 return status; 7007} 7008 7009void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7010{ 7011 for (size_t i = 0; i < mTracks.size(); ++i) { 7012 sp<Track> track = mTracks[i]; 7013 if (track->auxEffectId() == effectId) { 7014 attachAuxEffect_l(track, 0); 7015 } 7016 } 7017} 7018 7019status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7020{ 7021 // only one chain per input thread 7022 if (mEffectChains.size() != 0) { 7023 return INVALID_OPERATION; 7024 } 7025 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7026 7027 chain->setInBuffer(NULL); 7028 chain->setOutBuffer(NULL); 7029 7030 checkSuspendOnAddEffectChain_l(chain); 7031 7032 mEffectChains.add(chain); 7033 7034 return NO_ERROR; 7035} 7036 7037size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7038{ 7039 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7040 ALOGW_IF(mEffectChains.size() != 1, 7041 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7042 chain.get(), mEffectChains.size(), this); 7043 if (mEffectChains.size() == 1) { 7044 mEffectChains.removeAt(0); 7045 } 7046 return 0; 7047} 7048 7049// ---------------------------------------------------------------------------- 7050// EffectModule implementation 7051// ---------------------------------------------------------------------------- 7052 7053#undef LOG_TAG 7054#define LOG_TAG "AudioFlinger::EffectModule" 7055 7056AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7057 const wp<AudioFlinger::EffectChain>& chain, 7058 effect_descriptor_t *desc, 7059 int id, 7060 int sessionId) 7061 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7062 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7063{ 7064 ALOGV("Constructor %p", this); 7065 int lStatus; 7066 if (thread == NULL) { 7067 return; 7068 } 7069 7070 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7071 7072 // create effect engine from effect factory 7073 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7074 7075 if (mStatus != NO_ERROR) { 7076 return; 7077 } 7078 lStatus = init(); 7079 if (lStatus < 0) { 7080 mStatus = lStatus; 7081 goto Error; 7082 } 7083 7084 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7085 mPinned = true; 7086 } 7087 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7088 return; 7089Error: 7090 EffectRelease(mEffectInterface); 7091 mEffectInterface = NULL; 7092 ALOGV("Constructor Error %d", mStatus); 7093} 7094 7095AudioFlinger::EffectModule::~EffectModule() 7096{ 7097 ALOGV("Destructor %p", this); 7098 if (mEffectInterface != NULL) { 7099 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7100 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7101 sp<ThreadBase> thread = mThread.promote(); 7102 if (thread != 0) { 7103 audio_stream_t *stream = thread->stream(); 7104 if (stream != NULL) { 7105 stream->remove_audio_effect(stream, mEffectInterface); 7106 } 7107 } 7108 } 7109 // release effect engine 7110 EffectRelease(mEffectInterface); 7111 } 7112} 7113 7114status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7115{ 7116 status_t status; 7117 7118 Mutex::Autolock _l(mLock); 7119 int priority = handle->priority(); 7120 size_t size = mHandles.size(); 7121 sp<EffectHandle> h; 7122 size_t i; 7123 for (i = 0; i < size; i++) { 7124 h = mHandles[i].promote(); 7125 if (h == 0) continue; 7126 if (h->priority() <= priority) break; 7127 } 7128 // if inserted in first place, move effect control from previous owner to this handle 7129 if (i == 0) { 7130 bool enabled = false; 7131 if (h != 0) { 7132 enabled = h->enabled(); 7133 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7134 } 7135 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7136 status = NO_ERROR; 7137 } else { 7138 status = ALREADY_EXISTS; 7139 } 7140 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7141 mHandles.insertAt(handle, i); 7142 return status; 7143} 7144 7145size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7146{ 7147 Mutex::Autolock _l(mLock); 7148 size_t size = mHandles.size(); 7149 size_t i; 7150 for (i = 0; i < size; i++) { 7151 if (mHandles[i] == handle) break; 7152 } 7153 if (i == size) { 7154 return size; 7155 } 7156 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7157 7158 bool enabled = false; 7159 EffectHandle *hdl = handle.unsafe_get(); 7160 if (hdl != NULL) { 7161 ALOGV("removeHandle() unsafe_get OK"); 7162 enabled = hdl->enabled(); 7163 } 7164 mHandles.removeAt(i); 7165 size = mHandles.size(); 7166 // if removed from first place, move effect control from this handle to next in line 7167 if (i == 0 && size != 0) { 7168 sp<EffectHandle> h = mHandles[0].promote(); 7169 if (h != 0) { 7170 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7171 } 7172 } 7173 7174 // Prevent calls to process() and other functions on effect interface from now on. 7175 // The effect engine will be released by the destructor when the last strong reference on 7176 // this object is released which can happen after next process is called. 7177 if (size == 0 && !mPinned) { 7178 mState = DESTROYED; 7179 } 7180 7181 return size; 7182} 7183 7184sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7185{ 7186 Mutex::Autolock _l(mLock); 7187 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7188} 7189 7190void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7191{ 7192 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7193 // keep a strong reference on this EffectModule to avoid calling the 7194 // destructor before we exit 7195 sp<EffectModule> keep(this); 7196 { 7197 sp<ThreadBase> thread = mThread.promote(); 7198 if (thread != 0) { 7199 thread->disconnectEffect(keep, handle, unpinIfLast); 7200 } 7201 } 7202} 7203 7204void AudioFlinger::EffectModule::updateState() { 7205 Mutex::Autolock _l(mLock); 7206 7207 switch (mState) { 7208 case RESTART: 7209 reset_l(); 7210 // FALL THROUGH 7211 7212 case STARTING: 7213 // clear auxiliary effect input buffer for next accumulation 7214 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7215 memset(mConfig.inputCfg.buffer.raw, 7216 0, 7217 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7218 } 7219 start_l(); 7220 mState = ACTIVE; 7221 break; 7222 case STOPPING: 7223 stop_l(); 7224 mDisableWaitCnt = mMaxDisableWaitCnt; 7225 mState = STOPPED; 7226 break; 7227 case STOPPED: 7228 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7229 // turn off sequence. 7230 if (--mDisableWaitCnt == 0) { 7231 reset_l(); 7232 mState = IDLE; 7233 } 7234 break; 7235 default: //IDLE , ACTIVE, DESTROYED 7236 break; 7237 } 7238} 7239 7240void AudioFlinger::EffectModule::process() 7241{ 7242 Mutex::Autolock _l(mLock); 7243 7244 if (mState == DESTROYED || mEffectInterface == NULL || 7245 mConfig.inputCfg.buffer.raw == NULL || 7246 mConfig.outputCfg.buffer.raw == NULL) { 7247 return; 7248 } 7249 7250 if (isProcessEnabled()) { 7251 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7252 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7253 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7254 mConfig.inputCfg.buffer.s32, 7255 mConfig.inputCfg.buffer.frameCount/2); 7256 } 7257 7258 // do the actual processing in the effect engine 7259 int ret = (*mEffectInterface)->process(mEffectInterface, 7260 &mConfig.inputCfg.buffer, 7261 &mConfig.outputCfg.buffer); 7262 7263 // force transition to IDLE state when engine is ready 7264 if (mState == STOPPED && ret == -ENODATA) { 7265 mDisableWaitCnt = 1; 7266 } 7267 7268 // clear auxiliary effect input buffer for next accumulation 7269 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7270 memset(mConfig.inputCfg.buffer.raw, 0, 7271 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7272 } 7273 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7274 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7275 // If an insert effect is idle and input buffer is different from output buffer, 7276 // accumulate input onto output 7277 sp<EffectChain> chain = mChain.promote(); 7278 if (chain != 0 && chain->activeTrackCnt() != 0) { 7279 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7280 int16_t *in = mConfig.inputCfg.buffer.s16; 7281 int16_t *out = mConfig.outputCfg.buffer.s16; 7282 for (size_t i = 0; i < frameCnt; i++) { 7283 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7284 } 7285 } 7286 } 7287} 7288 7289void AudioFlinger::EffectModule::reset_l() 7290{ 7291 if (mEffectInterface == NULL) { 7292 return; 7293 } 7294 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7295} 7296 7297status_t AudioFlinger::EffectModule::configure() 7298{ 7299 uint32_t channels; 7300 if (mEffectInterface == NULL) { 7301 return NO_INIT; 7302 } 7303 7304 sp<ThreadBase> thread = mThread.promote(); 7305 if (thread == 0) { 7306 return DEAD_OBJECT; 7307 } 7308 7309 // TODO: handle configuration of effects replacing track process 7310 if (thread->channelCount() == 1) { 7311 channels = AUDIO_CHANNEL_OUT_MONO; 7312 } else { 7313 channels = AUDIO_CHANNEL_OUT_STEREO; 7314 } 7315 7316 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7317 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7318 } else { 7319 mConfig.inputCfg.channels = channels; 7320 } 7321 mConfig.outputCfg.channels = channels; 7322 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7323 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7324 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7325 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7326 mConfig.inputCfg.bufferProvider.cookie = NULL; 7327 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7328 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7329 mConfig.outputCfg.bufferProvider.cookie = NULL; 7330 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7331 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7332 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7333 // Insert effect: 7334 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7335 // always overwrites output buffer: input buffer == output buffer 7336 // - in other sessions: 7337 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7338 // other effect: overwrites output buffer: input buffer == output buffer 7339 // Auxiliary effect: 7340 // accumulates in output buffer: input buffer != output buffer 7341 // Therefore: accumulate <=> input buffer != output buffer 7342 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7343 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7344 } else { 7345 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7346 } 7347 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7348 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7349 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7350 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7351 7352 ALOGV("configure() %p thread %p buffer %p framecount %d", 7353 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7354 7355 status_t cmdStatus; 7356 uint32_t size = sizeof(int); 7357 status_t status = (*mEffectInterface)->command(mEffectInterface, 7358 EFFECT_CMD_SET_CONFIG, 7359 sizeof(effect_config_t), 7360 &mConfig, 7361 &size, 7362 &cmdStatus); 7363 if (status == 0) { 7364 status = cmdStatus; 7365 } 7366 7367 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7368 (1000 * mConfig.outputCfg.buffer.frameCount); 7369 7370 return status; 7371} 7372 7373status_t AudioFlinger::EffectModule::init() 7374{ 7375 Mutex::Autolock _l(mLock); 7376 if (mEffectInterface == NULL) { 7377 return NO_INIT; 7378 } 7379 status_t cmdStatus; 7380 uint32_t size = sizeof(status_t); 7381 status_t status = (*mEffectInterface)->command(mEffectInterface, 7382 EFFECT_CMD_INIT, 7383 0, 7384 NULL, 7385 &size, 7386 &cmdStatus); 7387 if (status == 0) { 7388 status = cmdStatus; 7389 } 7390 return status; 7391} 7392 7393status_t AudioFlinger::EffectModule::start() 7394{ 7395 Mutex::Autolock _l(mLock); 7396 return start_l(); 7397} 7398 7399status_t AudioFlinger::EffectModule::start_l() 7400{ 7401 if (mEffectInterface == NULL) { 7402 return NO_INIT; 7403 } 7404 status_t cmdStatus; 7405 uint32_t size = sizeof(status_t); 7406 status_t status = (*mEffectInterface)->command(mEffectInterface, 7407 EFFECT_CMD_ENABLE, 7408 0, 7409 NULL, 7410 &size, 7411 &cmdStatus); 7412 if (status == 0) { 7413 status = cmdStatus; 7414 } 7415 if (status == 0 && 7416 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7417 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7418 sp<ThreadBase> thread = mThread.promote(); 7419 if (thread != 0) { 7420 audio_stream_t *stream = thread->stream(); 7421 if (stream != NULL) { 7422 stream->add_audio_effect(stream, mEffectInterface); 7423 } 7424 } 7425 } 7426 return status; 7427} 7428 7429status_t AudioFlinger::EffectModule::stop() 7430{ 7431 Mutex::Autolock _l(mLock); 7432 return stop_l(); 7433} 7434 7435status_t AudioFlinger::EffectModule::stop_l() 7436{ 7437 if (mEffectInterface == NULL) { 7438 return NO_INIT; 7439 } 7440 status_t cmdStatus; 7441 uint32_t size = sizeof(status_t); 7442 status_t status = (*mEffectInterface)->command(mEffectInterface, 7443 EFFECT_CMD_DISABLE, 7444 0, 7445 NULL, 7446 &size, 7447 &cmdStatus); 7448 if (status == 0) { 7449 status = cmdStatus; 7450 } 7451 if (status == 0 && 7452 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7453 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7454 sp<ThreadBase> thread = mThread.promote(); 7455 if (thread != 0) { 7456 audio_stream_t *stream = thread->stream(); 7457 if (stream != NULL) { 7458 stream->remove_audio_effect(stream, mEffectInterface); 7459 } 7460 } 7461 } 7462 return status; 7463} 7464 7465status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7466 uint32_t cmdSize, 7467 void *pCmdData, 7468 uint32_t *replySize, 7469 void *pReplyData) 7470{ 7471 Mutex::Autolock _l(mLock); 7472// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7473 7474 if (mState == DESTROYED || mEffectInterface == NULL) { 7475 return NO_INIT; 7476 } 7477 status_t status = (*mEffectInterface)->command(mEffectInterface, 7478 cmdCode, 7479 cmdSize, 7480 pCmdData, 7481 replySize, 7482 pReplyData); 7483 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7484 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7485 for (size_t i = 1; i < mHandles.size(); i++) { 7486 sp<EffectHandle> h = mHandles[i].promote(); 7487 if (h != 0) { 7488 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7489 } 7490 } 7491 } 7492 return status; 7493} 7494 7495status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7496{ 7497 7498 Mutex::Autolock _l(mLock); 7499 ALOGV("setEnabled %p enabled %d", this, enabled); 7500 7501 if (enabled != isEnabled()) { 7502 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7503 if (enabled && status != NO_ERROR) { 7504 return status; 7505 } 7506 7507 switch (mState) { 7508 // going from disabled to enabled 7509 case IDLE: 7510 mState = STARTING; 7511 break; 7512 case STOPPED: 7513 mState = RESTART; 7514 break; 7515 case STOPPING: 7516 mState = ACTIVE; 7517 break; 7518 7519 // going from enabled to disabled 7520 case RESTART: 7521 mState = STOPPED; 7522 break; 7523 case STARTING: 7524 mState = IDLE; 7525 break; 7526 case ACTIVE: 7527 mState = STOPPING; 7528 break; 7529 case DESTROYED: 7530 return NO_ERROR; // simply ignore as we are being destroyed 7531 } 7532 for (size_t i = 1; i < mHandles.size(); i++) { 7533 sp<EffectHandle> h = mHandles[i].promote(); 7534 if (h != 0) { 7535 h->setEnabled(enabled); 7536 } 7537 } 7538 } 7539 return NO_ERROR; 7540} 7541 7542bool AudioFlinger::EffectModule::isEnabled() const 7543{ 7544 switch (mState) { 7545 case RESTART: 7546 case STARTING: 7547 case ACTIVE: 7548 return true; 7549 case IDLE: 7550 case STOPPING: 7551 case STOPPED: 7552 case DESTROYED: 7553 default: 7554 return false; 7555 } 7556} 7557 7558bool AudioFlinger::EffectModule::isProcessEnabled() const 7559{ 7560 switch (mState) { 7561 case RESTART: 7562 case ACTIVE: 7563 case STOPPING: 7564 case STOPPED: 7565 return true; 7566 case IDLE: 7567 case STARTING: 7568 case DESTROYED: 7569 default: 7570 return false; 7571 } 7572} 7573 7574status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7575{ 7576 Mutex::Autolock _l(mLock); 7577 status_t status = NO_ERROR; 7578 7579 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7580 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7581 if (isProcessEnabled() && 7582 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7583 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7584 status_t cmdStatus; 7585 uint32_t volume[2]; 7586 uint32_t *pVolume = NULL; 7587 uint32_t size = sizeof(volume); 7588 volume[0] = *left; 7589 volume[1] = *right; 7590 if (controller) { 7591 pVolume = volume; 7592 } 7593 status = (*mEffectInterface)->command(mEffectInterface, 7594 EFFECT_CMD_SET_VOLUME, 7595 size, 7596 volume, 7597 &size, 7598 pVolume); 7599 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7600 *left = volume[0]; 7601 *right = volume[1]; 7602 } 7603 } 7604 return status; 7605} 7606 7607status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7608{ 7609 Mutex::Autolock _l(mLock); 7610 status_t status = NO_ERROR; 7611 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7612 // audio pre processing modules on RecordThread can receive both output and 7613 // input device indication in the same call 7614 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7615 if (dev) { 7616 status_t cmdStatus; 7617 uint32_t size = sizeof(status_t); 7618 7619 status = (*mEffectInterface)->command(mEffectInterface, 7620 EFFECT_CMD_SET_DEVICE, 7621 sizeof(uint32_t), 7622 &dev, 7623 &size, 7624 &cmdStatus); 7625 if (status == NO_ERROR) { 7626 status = cmdStatus; 7627 } 7628 } 7629 dev = device & AUDIO_DEVICE_IN_ALL; 7630 if (dev) { 7631 status_t cmdStatus; 7632 uint32_t size = sizeof(status_t); 7633 7634 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7635 EFFECT_CMD_SET_INPUT_DEVICE, 7636 sizeof(uint32_t), 7637 &dev, 7638 &size, 7639 &cmdStatus); 7640 if (status2 == NO_ERROR) { 7641 status2 = cmdStatus; 7642 } 7643 if (status == NO_ERROR) { 7644 status = status2; 7645 } 7646 } 7647 } 7648 return status; 7649} 7650 7651status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7652{ 7653 Mutex::Autolock _l(mLock); 7654 status_t status = NO_ERROR; 7655 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7656 status_t cmdStatus; 7657 uint32_t size = sizeof(status_t); 7658 status = (*mEffectInterface)->command(mEffectInterface, 7659 EFFECT_CMD_SET_AUDIO_MODE, 7660 sizeof(audio_mode_t), 7661 &mode, 7662 &size, 7663 &cmdStatus); 7664 if (status == NO_ERROR) { 7665 status = cmdStatus; 7666 } 7667 } 7668 return status; 7669} 7670 7671void AudioFlinger::EffectModule::setSuspended(bool suspended) 7672{ 7673 Mutex::Autolock _l(mLock); 7674 mSuspended = suspended; 7675} 7676 7677bool AudioFlinger::EffectModule::suspended() const 7678{ 7679 Mutex::Autolock _l(mLock); 7680 return mSuspended; 7681} 7682 7683status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7684{ 7685 const size_t SIZE = 256; 7686 char buffer[SIZE]; 7687 String8 result; 7688 7689 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7690 result.append(buffer); 7691 7692 bool locked = tryLock(mLock); 7693 // failed to lock - AudioFlinger is probably deadlocked 7694 if (!locked) { 7695 result.append("\t\tCould not lock Fx mutex:\n"); 7696 } 7697 7698 result.append("\t\tSession Status State Engine:\n"); 7699 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7700 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7701 result.append(buffer); 7702 7703 result.append("\t\tDescriptor:\n"); 7704 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7705 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7706 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7707 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7708 result.append(buffer); 7709 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7710 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7711 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7712 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7713 result.append(buffer); 7714 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7715 mDescriptor.apiVersion, 7716 mDescriptor.flags); 7717 result.append(buffer); 7718 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7719 mDescriptor.name); 7720 result.append(buffer); 7721 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7722 mDescriptor.implementor); 7723 result.append(buffer); 7724 7725 result.append("\t\t- Input configuration:\n"); 7726 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7727 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7728 (uint32_t)mConfig.inputCfg.buffer.raw, 7729 mConfig.inputCfg.buffer.frameCount, 7730 mConfig.inputCfg.samplingRate, 7731 mConfig.inputCfg.channels, 7732 mConfig.inputCfg.format); 7733 result.append(buffer); 7734 7735 result.append("\t\t- Output configuration:\n"); 7736 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7737 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7738 (uint32_t)mConfig.outputCfg.buffer.raw, 7739 mConfig.outputCfg.buffer.frameCount, 7740 mConfig.outputCfg.samplingRate, 7741 mConfig.outputCfg.channels, 7742 mConfig.outputCfg.format); 7743 result.append(buffer); 7744 7745 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7746 result.append(buffer); 7747 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7748 for (size_t i = 0; i < mHandles.size(); ++i) { 7749 sp<EffectHandle> handle = mHandles[i].promote(); 7750 if (handle != 0) { 7751 handle->dump(buffer, SIZE); 7752 result.append(buffer); 7753 } 7754 } 7755 7756 result.append("\n"); 7757 7758 write(fd, result.string(), result.length()); 7759 7760 if (locked) { 7761 mLock.unlock(); 7762 } 7763 7764 return NO_ERROR; 7765} 7766 7767// ---------------------------------------------------------------------------- 7768// EffectHandle implementation 7769// ---------------------------------------------------------------------------- 7770 7771#undef LOG_TAG 7772#define LOG_TAG "AudioFlinger::EffectHandle" 7773 7774AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7775 const sp<AudioFlinger::Client>& client, 7776 const sp<IEffectClient>& effectClient, 7777 int32_t priority) 7778 : BnEffect(), 7779 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7780 mPriority(priority), mHasControl(false), mEnabled(false) 7781{ 7782 ALOGV("constructor %p", this); 7783 7784 if (client == 0) { 7785 return; 7786 } 7787 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7788 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7789 if (mCblkMemory != 0) { 7790 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7791 7792 if (mCblk != NULL) { 7793 new(mCblk) effect_param_cblk_t(); 7794 mBuffer = (uint8_t *)mCblk + bufOffset; 7795 } 7796 } else { 7797 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7798 return; 7799 } 7800} 7801 7802AudioFlinger::EffectHandle::~EffectHandle() 7803{ 7804 ALOGV("Destructor %p", this); 7805 disconnect(false); 7806 ALOGV("Destructor DONE %p", this); 7807} 7808 7809status_t AudioFlinger::EffectHandle::enable() 7810{ 7811 ALOGV("enable %p", this); 7812 if (!mHasControl) return INVALID_OPERATION; 7813 if (mEffect == 0) return DEAD_OBJECT; 7814 7815 if (mEnabled) { 7816 return NO_ERROR; 7817 } 7818 7819 mEnabled = true; 7820 7821 sp<ThreadBase> thread = mEffect->thread().promote(); 7822 if (thread != 0) { 7823 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7824 } 7825 7826 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7827 if (mEffect->suspended()) { 7828 return NO_ERROR; 7829 } 7830 7831 status_t status = mEffect->setEnabled(true); 7832 if (status != NO_ERROR) { 7833 if (thread != 0) { 7834 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7835 } 7836 mEnabled = false; 7837 } 7838 return status; 7839} 7840 7841status_t AudioFlinger::EffectHandle::disable() 7842{ 7843 ALOGV("disable %p", this); 7844 if (!mHasControl) return INVALID_OPERATION; 7845 if (mEffect == 0) return DEAD_OBJECT; 7846 7847 if (!mEnabled) { 7848 return NO_ERROR; 7849 } 7850 mEnabled = false; 7851 7852 if (mEffect->suspended()) { 7853 return NO_ERROR; 7854 } 7855 7856 status_t status = mEffect->setEnabled(false); 7857 7858 sp<ThreadBase> thread = mEffect->thread().promote(); 7859 if (thread != 0) { 7860 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7861 } 7862 7863 return status; 7864} 7865 7866void AudioFlinger::EffectHandle::disconnect() 7867{ 7868 disconnect(true); 7869} 7870 7871void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7872{ 7873 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7874 if (mEffect == 0) { 7875 return; 7876 } 7877 mEffect->disconnect(this, unpinIfLast); 7878 7879 if (mHasControl && mEnabled) { 7880 sp<ThreadBase> thread = mEffect->thread().promote(); 7881 if (thread != 0) { 7882 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7883 } 7884 } 7885 7886 // release sp on module => module destructor can be called now 7887 mEffect.clear(); 7888 if (mClient != 0) { 7889 if (mCblk != NULL) { 7890 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7891 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7892 } 7893 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7894 // Client destructor must run with AudioFlinger mutex locked 7895 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7896 mClient.clear(); 7897 } 7898} 7899 7900status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7901 uint32_t cmdSize, 7902 void *pCmdData, 7903 uint32_t *replySize, 7904 void *pReplyData) 7905{ 7906// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7907// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7908 7909 // only get parameter command is permitted for applications not controlling the effect 7910 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7911 return INVALID_OPERATION; 7912 } 7913 if (mEffect == 0) return DEAD_OBJECT; 7914 if (mClient == 0) return INVALID_OPERATION; 7915 7916 // handle commands that are not forwarded transparently to effect engine 7917 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7918 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7919 // no risk to block the whole media server process or mixer threads is we are stuck here 7920 Mutex::Autolock _l(mCblk->lock); 7921 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7922 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7923 mCblk->serverIndex = 0; 7924 mCblk->clientIndex = 0; 7925 return BAD_VALUE; 7926 } 7927 status_t status = NO_ERROR; 7928 while (mCblk->serverIndex < mCblk->clientIndex) { 7929 int reply; 7930 uint32_t rsize = sizeof(int); 7931 int *p = (int *)(mBuffer + mCblk->serverIndex); 7932 int size = *p++; 7933 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7934 ALOGW("command(): invalid parameter block size"); 7935 break; 7936 } 7937 effect_param_t *param = (effect_param_t *)p; 7938 if (param->psize == 0 || param->vsize == 0) { 7939 ALOGW("command(): null parameter or value size"); 7940 mCblk->serverIndex += size; 7941 continue; 7942 } 7943 uint32_t psize = sizeof(effect_param_t) + 7944 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7945 param->vsize; 7946 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7947 psize, 7948 p, 7949 &rsize, 7950 &reply); 7951 // stop at first error encountered 7952 if (ret != NO_ERROR) { 7953 status = ret; 7954 *(int *)pReplyData = reply; 7955 break; 7956 } else if (reply != NO_ERROR) { 7957 *(int *)pReplyData = reply; 7958 break; 7959 } 7960 mCblk->serverIndex += size; 7961 } 7962 mCblk->serverIndex = 0; 7963 mCblk->clientIndex = 0; 7964 return status; 7965 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7966 *(int *)pReplyData = NO_ERROR; 7967 return enable(); 7968 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7969 *(int *)pReplyData = NO_ERROR; 7970 return disable(); 7971 } 7972 7973 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7974} 7975 7976void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7977{ 7978 ALOGV("setControl %p control %d", this, hasControl); 7979 7980 mHasControl = hasControl; 7981 mEnabled = enabled; 7982 7983 if (signal && mEffectClient != 0) { 7984 mEffectClient->controlStatusChanged(hasControl); 7985 } 7986} 7987 7988void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7989 uint32_t cmdSize, 7990 void *pCmdData, 7991 uint32_t replySize, 7992 void *pReplyData) 7993{ 7994 if (mEffectClient != 0) { 7995 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7996 } 7997} 7998 7999 8000 8001void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8002{ 8003 if (mEffectClient != 0) { 8004 mEffectClient->enableStatusChanged(enabled); 8005 } 8006} 8007 8008status_t AudioFlinger::EffectHandle::onTransact( 8009 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8010{ 8011 return BnEffect::onTransact(code, data, reply, flags); 8012} 8013 8014 8015void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8016{ 8017 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8018 8019 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8020 (mClient == 0) ? getpid_cached : mClient->pid(), 8021 mPriority, 8022 mHasControl, 8023 !locked, 8024 mCblk ? mCblk->clientIndex : 0, 8025 mCblk ? mCblk->serverIndex : 0 8026 ); 8027 8028 if (locked) { 8029 mCblk->lock.unlock(); 8030 } 8031} 8032 8033#undef LOG_TAG 8034#define LOG_TAG "AudioFlinger::EffectChain" 8035 8036AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8037 int sessionId) 8038 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8039 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8040 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8041{ 8042 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8043 if (thread == NULL) { 8044 return; 8045 } 8046 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8047 thread->frameCount(); 8048} 8049 8050AudioFlinger::EffectChain::~EffectChain() 8051{ 8052 if (mOwnInBuffer) { 8053 delete mInBuffer; 8054 } 8055 8056} 8057 8058// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8059sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8060{ 8061 size_t size = mEffects.size(); 8062 8063 for (size_t i = 0; i < size; i++) { 8064 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8065 return mEffects[i]; 8066 } 8067 } 8068 return 0; 8069} 8070 8071// getEffectFromId_l() must be called with ThreadBase::mLock held 8072sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8073{ 8074 size_t size = mEffects.size(); 8075 8076 for (size_t i = 0; i < size; i++) { 8077 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8078 if (id == 0 || mEffects[i]->id() == id) { 8079 return mEffects[i]; 8080 } 8081 } 8082 return 0; 8083} 8084 8085// getEffectFromType_l() must be called with ThreadBase::mLock held 8086sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8087 const effect_uuid_t *type) 8088{ 8089 size_t size = mEffects.size(); 8090 8091 for (size_t i = 0; i < size; i++) { 8092 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8093 return mEffects[i]; 8094 } 8095 } 8096 return 0; 8097} 8098 8099// Must be called with EffectChain::mLock locked 8100void AudioFlinger::EffectChain::process_l() 8101{ 8102 sp<ThreadBase> thread = mThread.promote(); 8103 if (thread == 0) { 8104 ALOGW("process_l(): cannot promote mixer thread"); 8105 return; 8106 } 8107 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8108 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8109 // always process effects unless no more tracks are on the session and the effect tail 8110 // has been rendered 8111 bool doProcess = true; 8112 if (!isGlobalSession) { 8113 bool tracksOnSession = (trackCnt() != 0); 8114 8115 if (!tracksOnSession && mTailBufferCount == 0) { 8116 doProcess = false; 8117 } 8118 8119 if (activeTrackCnt() == 0) { 8120 // if no track is active and the effect tail has not been rendered, 8121 // the input buffer must be cleared here as the mixer process will not do it 8122 if (tracksOnSession || mTailBufferCount > 0) { 8123 size_t numSamples = thread->frameCount() * thread->channelCount(); 8124 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8125 if (mTailBufferCount > 0) { 8126 mTailBufferCount--; 8127 } 8128 } 8129 } 8130 } 8131 8132 size_t size = mEffects.size(); 8133 if (doProcess) { 8134 for (size_t i = 0; i < size; i++) { 8135 mEffects[i]->process(); 8136 } 8137 } 8138 for (size_t i = 0; i < size; i++) { 8139 mEffects[i]->updateState(); 8140 } 8141} 8142 8143// addEffect_l() must be called with PlaybackThread::mLock held 8144status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8145{ 8146 effect_descriptor_t desc = effect->desc(); 8147 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8148 8149 Mutex::Autolock _l(mLock); 8150 effect->setChain(this); 8151 sp<ThreadBase> thread = mThread.promote(); 8152 if (thread == 0) { 8153 return NO_INIT; 8154 } 8155 effect->setThread(thread); 8156 8157 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8158 // Auxiliary effects are inserted at the beginning of mEffects vector as 8159 // they are processed first and accumulated in chain input buffer 8160 mEffects.insertAt(effect, 0); 8161 8162 // the input buffer for auxiliary effect contains mono samples in 8163 // 32 bit format. This is to avoid saturation in AudoMixer 8164 // accumulation stage. Saturation is done in EffectModule::process() before 8165 // calling the process in effect engine 8166 size_t numSamples = thread->frameCount(); 8167 int32_t *buffer = new int32_t[numSamples]; 8168 memset(buffer, 0, numSamples * sizeof(int32_t)); 8169 effect->setInBuffer((int16_t *)buffer); 8170 // auxiliary effects output samples to chain input buffer for further processing 8171 // by insert effects 8172 effect->setOutBuffer(mInBuffer); 8173 } else { 8174 // Insert effects are inserted at the end of mEffects vector as they are processed 8175 // after track and auxiliary effects. 8176 // Insert effect order as a function of indicated preference: 8177 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8178 // another effect is present 8179 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8180 // last effect claiming first position 8181 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8182 // first effect claiming last position 8183 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8184 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8185 // already present 8186 8187 size_t size = mEffects.size(); 8188 size_t idx_insert = size; 8189 ssize_t idx_insert_first = -1; 8190 ssize_t idx_insert_last = -1; 8191 8192 for (size_t i = 0; i < size; i++) { 8193 effect_descriptor_t d = mEffects[i]->desc(); 8194 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8195 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8196 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8197 // check invalid effect chaining combinations 8198 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8199 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8200 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8201 return INVALID_OPERATION; 8202 } 8203 // remember position of first insert effect and by default 8204 // select this as insert position for new effect 8205 if (idx_insert == size) { 8206 idx_insert = i; 8207 } 8208 // remember position of last insert effect claiming 8209 // first position 8210 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8211 idx_insert_first = i; 8212 } 8213 // remember position of first insert effect claiming 8214 // last position 8215 if (iPref == EFFECT_FLAG_INSERT_LAST && 8216 idx_insert_last == -1) { 8217 idx_insert_last = i; 8218 } 8219 } 8220 } 8221 8222 // modify idx_insert from first position if needed 8223 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8224 if (idx_insert_last != -1) { 8225 idx_insert = idx_insert_last; 8226 } else { 8227 idx_insert = size; 8228 } 8229 } else { 8230 if (idx_insert_first != -1) { 8231 idx_insert = idx_insert_first + 1; 8232 } 8233 } 8234 8235 // always read samples from chain input buffer 8236 effect->setInBuffer(mInBuffer); 8237 8238 // if last effect in the chain, output samples to chain 8239 // output buffer, otherwise to chain input buffer 8240 if (idx_insert == size) { 8241 if (idx_insert != 0) { 8242 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8243 mEffects[idx_insert-1]->configure(); 8244 } 8245 effect->setOutBuffer(mOutBuffer); 8246 } else { 8247 effect->setOutBuffer(mInBuffer); 8248 } 8249 mEffects.insertAt(effect, idx_insert); 8250 8251 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8252 } 8253 effect->configure(); 8254 return NO_ERROR; 8255} 8256 8257// removeEffect_l() must be called with PlaybackThread::mLock held 8258size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8259{ 8260 Mutex::Autolock _l(mLock); 8261 size_t size = mEffects.size(); 8262 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8263 8264 for (size_t i = 0; i < size; i++) { 8265 if (effect == mEffects[i]) { 8266 // calling stop here will remove pre-processing effect from the audio HAL. 8267 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8268 // the middle of a read from audio HAL 8269 if (mEffects[i]->state() == EffectModule::ACTIVE || 8270 mEffects[i]->state() == EffectModule::STOPPING) { 8271 mEffects[i]->stop(); 8272 } 8273 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8274 delete[] effect->inBuffer(); 8275 } else { 8276 if (i == size - 1 && i != 0) { 8277 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8278 mEffects[i - 1]->configure(); 8279 } 8280 } 8281 mEffects.removeAt(i); 8282 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8283 break; 8284 } 8285 } 8286 8287 return mEffects.size(); 8288} 8289 8290// setDevice_l() must be called with PlaybackThread::mLock held 8291void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8292{ 8293 size_t size = mEffects.size(); 8294 for (size_t i = 0; i < size; i++) { 8295 mEffects[i]->setDevice(device); 8296 } 8297} 8298 8299// setMode_l() must be called with PlaybackThread::mLock held 8300void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8301{ 8302 size_t size = mEffects.size(); 8303 for (size_t i = 0; i < size; i++) { 8304 mEffects[i]->setMode(mode); 8305 } 8306} 8307 8308// setVolume_l() must be called with PlaybackThread::mLock held 8309bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8310{ 8311 uint32_t newLeft = *left; 8312 uint32_t newRight = *right; 8313 bool hasControl = false; 8314 int ctrlIdx = -1; 8315 size_t size = mEffects.size(); 8316 8317 // first update volume controller 8318 for (size_t i = size; i > 0; i--) { 8319 if (mEffects[i - 1]->isProcessEnabled() && 8320 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8321 ctrlIdx = i - 1; 8322 hasControl = true; 8323 break; 8324 } 8325 } 8326 8327 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8328 if (hasControl) { 8329 *left = mNewLeftVolume; 8330 *right = mNewRightVolume; 8331 } 8332 return hasControl; 8333 } 8334 8335 mVolumeCtrlIdx = ctrlIdx; 8336 mLeftVolume = newLeft; 8337 mRightVolume = newRight; 8338 8339 // second get volume update from volume controller 8340 if (ctrlIdx >= 0) { 8341 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8342 mNewLeftVolume = newLeft; 8343 mNewRightVolume = newRight; 8344 } 8345 // then indicate volume to all other effects in chain. 8346 // Pass altered volume to effects before volume controller 8347 // and requested volume to effects after controller 8348 uint32_t lVol = newLeft; 8349 uint32_t rVol = newRight; 8350 8351 for (size_t i = 0; i < size; i++) { 8352 if ((int)i == ctrlIdx) continue; 8353 // this also works for ctrlIdx == -1 when there is no volume controller 8354 if ((int)i > ctrlIdx) { 8355 lVol = *left; 8356 rVol = *right; 8357 } 8358 mEffects[i]->setVolume(&lVol, &rVol, false); 8359 } 8360 *left = newLeft; 8361 *right = newRight; 8362 8363 return hasControl; 8364} 8365 8366status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8367{ 8368 const size_t SIZE = 256; 8369 char buffer[SIZE]; 8370 String8 result; 8371 8372 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8373 result.append(buffer); 8374 8375 bool locked = tryLock(mLock); 8376 // failed to lock - AudioFlinger is probably deadlocked 8377 if (!locked) { 8378 result.append("\tCould not lock mutex:\n"); 8379 } 8380 8381 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8382 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8383 mEffects.size(), 8384 (uint32_t)mInBuffer, 8385 (uint32_t)mOutBuffer, 8386 mActiveTrackCnt); 8387 result.append(buffer); 8388 write(fd, result.string(), result.size()); 8389 8390 for (size_t i = 0; i < mEffects.size(); ++i) { 8391 sp<EffectModule> effect = mEffects[i]; 8392 if (effect != 0) { 8393 effect->dump(fd, args); 8394 } 8395 } 8396 8397 if (locked) { 8398 mLock.unlock(); 8399 } 8400 8401 return NO_ERROR; 8402} 8403 8404// must be called with ThreadBase::mLock held 8405void AudioFlinger::EffectChain::setEffectSuspended_l( 8406 const effect_uuid_t *type, bool suspend) 8407{ 8408 sp<SuspendedEffectDesc> desc; 8409 // use effect type UUID timelow as key as there is no real risk of identical 8410 // timeLow fields among effect type UUIDs. 8411 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8412 if (suspend) { 8413 if (index >= 0) { 8414 desc = mSuspendedEffects.valueAt(index); 8415 } else { 8416 desc = new SuspendedEffectDesc(); 8417 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8418 mSuspendedEffects.add(type->timeLow, desc); 8419 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8420 } 8421 if (desc->mRefCount++ == 0) { 8422 sp<EffectModule> effect = getEffectIfEnabled(type); 8423 if (effect != 0) { 8424 desc->mEffect = effect; 8425 effect->setSuspended(true); 8426 effect->setEnabled(false); 8427 } 8428 } 8429 } else { 8430 if (index < 0) { 8431 return; 8432 } 8433 desc = mSuspendedEffects.valueAt(index); 8434 if (desc->mRefCount <= 0) { 8435 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8436 desc->mRefCount = 1; 8437 } 8438 if (--desc->mRefCount == 0) { 8439 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8440 if (desc->mEffect != 0) { 8441 sp<EffectModule> effect = desc->mEffect.promote(); 8442 if (effect != 0) { 8443 effect->setSuspended(false); 8444 sp<EffectHandle> handle = effect->controlHandle(); 8445 if (handle != 0) { 8446 effect->setEnabled(handle->enabled()); 8447 } 8448 } 8449 desc->mEffect.clear(); 8450 } 8451 mSuspendedEffects.removeItemsAt(index); 8452 } 8453 } 8454} 8455 8456// must be called with ThreadBase::mLock held 8457void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8458{ 8459 sp<SuspendedEffectDesc> desc; 8460 8461 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8462 if (suspend) { 8463 if (index >= 0) { 8464 desc = mSuspendedEffects.valueAt(index); 8465 } else { 8466 desc = new SuspendedEffectDesc(); 8467 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8468 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8469 } 8470 if (desc->mRefCount++ == 0) { 8471 Vector< sp<EffectModule> > effects; 8472 getSuspendEligibleEffects(effects); 8473 for (size_t i = 0; i < effects.size(); i++) { 8474 setEffectSuspended_l(&effects[i]->desc().type, true); 8475 } 8476 } 8477 } else { 8478 if (index < 0) { 8479 return; 8480 } 8481 desc = mSuspendedEffects.valueAt(index); 8482 if (desc->mRefCount <= 0) { 8483 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8484 desc->mRefCount = 1; 8485 } 8486 if (--desc->mRefCount == 0) { 8487 Vector<const effect_uuid_t *> types; 8488 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8489 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8490 continue; 8491 } 8492 types.add(&mSuspendedEffects.valueAt(i)->mType); 8493 } 8494 for (size_t i = 0; i < types.size(); i++) { 8495 setEffectSuspended_l(types[i], false); 8496 } 8497 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8498 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8499 } 8500 } 8501} 8502 8503 8504// The volume effect is used for automated tests only 8505#ifndef OPENSL_ES_H_ 8506static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8507 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8508const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8509#endif //OPENSL_ES_H_ 8510 8511bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8512{ 8513 // auxiliary effects and visualizer are never suspended on output mix 8514 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8515 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8516 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8517 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8518 return false; 8519 } 8520 return true; 8521} 8522 8523void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8524{ 8525 effects.clear(); 8526 for (size_t i = 0; i < mEffects.size(); i++) { 8527 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8528 effects.add(mEffects[i]); 8529 } 8530 } 8531} 8532 8533sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8534 const effect_uuid_t *type) 8535{ 8536 sp<EffectModule> effect = getEffectFromType_l(type); 8537 return effect != 0 && effect->isEnabled() ? effect : 0; 8538} 8539 8540void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8541 bool enabled) 8542{ 8543 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8544 if (enabled) { 8545 if (index < 0) { 8546 // if the effect is not suspend check if all effects are suspended 8547 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8548 if (index < 0) { 8549 return; 8550 } 8551 if (!isEffectEligibleForSuspend(effect->desc())) { 8552 return; 8553 } 8554 setEffectSuspended_l(&effect->desc().type, enabled); 8555 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8556 if (index < 0) { 8557 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8558 return; 8559 } 8560 } 8561 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8562 effect->desc().type.timeLow); 8563 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8564 // if effect is requested to suspended but was not yet enabled, supend it now. 8565 if (desc->mEffect == 0) { 8566 desc->mEffect = effect; 8567 effect->setEnabled(false); 8568 effect->setSuspended(true); 8569 } 8570 } else { 8571 if (index < 0) { 8572 return; 8573 } 8574 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8575 effect->desc().type.timeLow); 8576 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8577 desc->mEffect.clear(); 8578 effect->setSuspended(false); 8579 } 8580} 8581 8582#undef LOG_TAG 8583#define LOG_TAG "AudioFlinger" 8584 8585// ---------------------------------------------------------------------------- 8586 8587status_t AudioFlinger::onTransact( 8588 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8589{ 8590 return BnAudioFlinger::onTransact(code, data, reply, flags); 8591} 8592 8593}; // namespace android 8594