AudioFlinger.cpp revision 0ca3cf94c0dfc173ad7886ae162c4b67067539f6
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 int *sessionId, 405 status_t *status) 406{ 407 sp<PlaybackThread::Track> track; 408 sp<TrackHandle> trackHandle; 409 sp<Client> client; 410 status_t lStatus; 411 int lSessionId; 412 413 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 414 // but if someone uses binder directly they could bypass that and cause us to crash 415 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 416 ALOGE("createTrack() invalid stream type %d", streamType); 417 lStatus = BAD_VALUE; 418 goto Exit; 419 } 420 421 { 422 Mutex::Autolock _l(mLock); 423 PlaybackThread *thread = checkPlaybackThread_l(output); 424 PlaybackThread *effectThread = NULL; 425 if (thread == NULL) { 426 ALOGE("unknown output thread"); 427 lStatus = BAD_VALUE; 428 goto Exit; 429 } 430 431 client = registerPid_l(pid); 432 433 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 434 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 435 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 436 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 437 if (mPlaybackThreads.keyAt(i) != output) { 438 // prevent same audio session on different output threads 439 uint32_t sessions = t->hasAudioSession(*sessionId); 440 if (sessions & PlaybackThread::TRACK_SESSION) { 441 ALOGE("createTrack() session ID %d already in use", *sessionId); 442 lStatus = BAD_VALUE; 443 goto Exit; 444 } 445 // check if an effect with same session ID is waiting for a track to be created 446 if (sessions & PlaybackThread::EFFECT_SESSION) { 447 effectThread = t.get(); 448 } 449 } 450 } 451 lSessionId = *sessionId; 452 } else { 453 // if no audio session id is provided, create one here 454 lSessionId = nextUniqueId(); 455 if (sessionId != NULL) { 456 *sessionId = lSessionId; 457 } 458 } 459 ALOGV("createTrack() lSessionId: %d", lSessionId); 460 461 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 462 track = thread->createTrack_l(client, streamType, sampleRate, format, 463 channelMask, frameCount, sharedBuffer, lSessionId, flags, &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 status_t *status) 1573{ 1574 sp<Track> track; 1575 status_t lStatus; 1576 1577 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1578 1579 // client expresses a preference for FAST, but we get the final say 1580 if ((flags & IAudioFlinger::TRACK_FAST) && 1581 !( 1582 // not timed 1583 (!isTimed) && 1584 // either of these use cases: 1585 ( 1586 // use case 1: shared buffer with any frame count 1587 ( 1588 (sharedBuffer != 0) 1589 ) || 1590 // use case 2: callback handler and small power-of-2 frame count 1591 ( 1592 // unfortunately we can't verify that there's a callback until start() 1593 // FIXME supported frame counts should not be hard-coded 1594 ( 1595 (frameCount == 128) || 1596 (frameCount == 256) || 1597 (frameCount == 512) 1598 ) 1599 ) 1600 ) && 1601 // PCM data 1602 audio_is_linear_pcm(format) && 1603 // mono or stereo 1604 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1605 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1606 // hardware sample rate 1607 (sampleRate == mSampleRate) 1608 // FIXME test that MixerThread for this fast track has a capable output HAL 1609 // FIXME add a permission test also? 1610 ) ) { 1611 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied"); 1612 flags &= ~IAudioFlinger::TRACK_FAST; 1613 } 1614 1615 if (mType == DIRECT) { 1616 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1617 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1618 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1619 "for output %p with format %d", 1620 sampleRate, format, channelMask, mOutput, mFormat); 1621 lStatus = BAD_VALUE; 1622 goto Exit; 1623 } 1624 } 1625 } else { 1626 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1627 if (sampleRate > mSampleRate*2) { 1628 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1629 lStatus = BAD_VALUE; 1630 goto Exit; 1631 } 1632 } 1633 1634 lStatus = initCheck(); 1635 if (lStatus != NO_ERROR) { 1636 ALOGE("Audio driver not initialized."); 1637 goto Exit; 1638 } 1639 1640 { // scope for mLock 1641 Mutex::Autolock _l(mLock); 1642 1643 // all tracks in same audio session must share the same routing strategy otherwise 1644 // conflicts will happen when tracks are moved from one output to another by audio policy 1645 // manager 1646 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1647 for (size_t i = 0; i < mTracks.size(); ++i) { 1648 sp<Track> t = mTracks[i]; 1649 if (t != 0 && !t->isOutputTrack()) { 1650 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1651 if (sessionId == t->sessionId() && strategy != actual) { 1652 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1653 strategy, actual); 1654 lStatus = BAD_VALUE; 1655 goto Exit; 1656 } 1657 } 1658 } 1659 1660 if (!isTimed) { 1661 track = new Track(this, client, streamType, sampleRate, format, 1662 channelMask, frameCount, sharedBuffer, sessionId, flags); 1663 } else { 1664 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1665 channelMask, frameCount, sharedBuffer, sessionId); 1666 } 1667 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1668 lStatus = NO_MEMORY; 1669 goto Exit; 1670 } 1671 mTracks.add(track); 1672 1673 sp<EffectChain> chain = getEffectChain_l(sessionId); 1674 if (chain != 0) { 1675 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1676 track->setMainBuffer(chain->inBuffer()); 1677 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1678 chain->incTrackCnt(); 1679 } 1680 } 1681 lStatus = NO_ERROR; 1682 1683Exit: 1684 if (status) { 1685 *status = lStatus; 1686 } 1687 return track; 1688} 1689 1690uint32_t AudioFlinger::PlaybackThread::latency() const 1691{ 1692 Mutex::Autolock _l(mLock); 1693 if (initCheck() == NO_ERROR) { 1694 return mOutput->stream->get_latency(mOutput->stream); 1695 } else { 1696 return 0; 1697 } 1698} 1699 1700void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1701{ 1702 Mutex::Autolock _l(mLock); 1703 mMasterVolume = value; 1704} 1705 1706void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1707{ 1708 Mutex::Autolock _l(mLock); 1709 setMasterMute_l(muted); 1710} 1711 1712void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1713{ 1714 Mutex::Autolock _l(mLock); 1715 mStreamTypes[stream].volume = value; 1716} 1717 1718void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1719{ 1720 Mutex::Autolock _l(mLock); 1721 mStreamTypes[stream].mute = muted; 1722} 1723 1724float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1725{ 1726 Mutex::Autolock _l(mLock); 1727 return mStreamTypes[stream].volume; 1728} 1729 1730// addTrack_l() must be called with ThreadBase::mLock held 1731status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1732{ 1733 status_t status = ALREADY_EXISTS; 1734 1735 // set retry count for buffer fill 1736 track->mRetryCount = kMaxTrackStartupRetries; 1737 if (mActiveTracks.indexOf(track) < 0) { 1738 // the track is newly added, make sure it fills up all its 1739 // buffers before playing. This is to ensure the client will 1740 // effectively get the latency it requested. 1741 track->mFillingUpStatus = Track::FS_FILLING; 1742 track->mResetDone = false; 1743 mActiveTracks.add(track); 1744 if (track->mainBuffer() != mMixBuffer) { 1745 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1746 if (chain != 0) { 1747 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1748 chain->incActiveTrackCnt(); 1749 } 1750 } 1751 1752 status = NO_ERROR; 1753 } 1754 1755 ALOGV("mWaitWorkCV.broadcast"); 1756 mWaitWorkCV.broadcast(); 1757 1758 return status; 1759} 1760 1761// destroyTrack_l() must be called with ThreadBase::mLock held 1762void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1763{ 1764 track->mState = TrackBase::TERMINATED; 1765 if (mActiveTracks.indexOf(track) < 0) { 1766 removeTrack_l(track); 1767 } 1768} 1769 1770void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1771{ 1772 mTracks.remove(track); 1773 deleteTrackName_l(track->name()); 1774 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1775 if (chain != 0) { 1776 chain->decTrackCnt(); 1777 } 1778} 1779 1780String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1781{ 1782 String8 out_s8 = String8(""); 1783 char *s; 1784 1785 Mutex::Autolock _l(mLock); 1786 if (initCheck() != NO_ERROR) { 1787 return out_s8; 1788 } 1789 1790 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1791 out_s8 = String8(s); 1792 free(s); 1793 return out_s8; 1794} 1795 1796// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1797void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1798 AudioSystem::OutputDescriptor desc; 1799 void *param2 = NULL; 1800 1801 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1802 1803 switch (event) { 1804 case AudioSystem::OUTPUT_OPENED: 1805 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1806 desc.channels = mChannelMask; 1807 desc.samplingRate = mSampleRate; 1808 desc.format = mFormat; 1809 desc.frameCount = mFrameCount; 1810 desc.latency = latency(); 1811 param2 = &desc; 1812 break; 1813 1814 case AudioSystem::STREAM_CONFIG_CHANGED: 1815 param2 = ¶m; 1816 case AudioSystem::OUTPUT_CLOSED: 1817 default: 1818 break; 1819 } 1820 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1821} 1822 1823void AudioFlinger::PlaybackThread::readOutputParameters() 1824{ 1825 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1826 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1827 mChannelCount = (uint16_t)popcount(mChannelMask); 1828 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1829 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1830 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1831 1832 // FIXME - Current mixer implementation only supports stereo output: Always 1833 // Allocate a stereo buffer even if HW output is mono. 1834 delete[] mMixBuffer; 1835 mMixBuffer = new int16_t[mFrameCount * 2]; 1836 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1837 1838 // force reconfiguration of effect chains and engines to take new buffer size and audio 1839 // parameters into account 1840 // Note that mLock is not held when readOutputParameters() is called from the constructor 1841 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1842 // matter. 1843 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1844 Vector< sp<EffectChain> > effectChains = mEffectChains; 1845 for (size_t i = 0; i < effectChains.size(); i ++) { 1846 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1847 } 1848} 1849 1850status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1851{ 1852 if (halFrames == NULL || dspFrames == NULL) { 1853 return BAD_VALUE; 1854 } 1855 Mutex::Autolock _l(mLock); 1856 if (initCheck() != NO_ERROR) { 1857 return INVALID_OPERATION; 1858 } 1859 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1860 1861 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1862} 1863 1864uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1865{ 1866 Mutex::Autolock _l(mLock); 1867 uint32_t result = 0; 1868 if (getEffectChain_l(sessionId) != 0) { 1869 result = EFFECT_SESSION; 1870 } 1871 1872 for (size_t i = 0; i < mTracks.size(); ++i) { 1873 sp<Track> track = mTracks[i]; 1874 if (sessionId == track->sessionId() && 1875 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1876 result |= TRACK_SESSION; 1877 break; 1878 } 1879 } 1880 1881 return result; 1882} 1883 1884uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1885{ 1886 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1887 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1888 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1889 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1890 } 1891 for (size_t i = 0; i < mTracks.size(); i++) { 1892 sp<Track> track = mTracks[i]; 1893 if (sessionId == track->sessionId() && 1894 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1895 return AudioSystem::getStrategyForStream(track->streamType()); 1896 } 1897 } 1898 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1899} 1900 1901 1902AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1903{ 1904 Mutex::Autolock _l(mLock); 1905 return mOutput; 1906} 1907 1908AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1909{ 1910 Mutex::Autolock _l(mLock); 1911 AudioStreamOut *output = mOutput; 1912 mOutput = NULL; 1913 return output; 1914} 1915 1916// this method must always be called either with ThreadBase mLock held or inside the thread loop 1917audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1918{ 1919 if (mOutput == NULL) { 1920 return NULL; 1921 } 1922 return &mOutput->stream->common; 1923} 1924 1925uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1926{ 1927 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1928 // decoding and transfer time. So sleeping for half of the latency would likely cause 1929 // underruns 1930 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1931 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1932 } else { 1933 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1934 } 1935} 1936 1937status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1938{ 1939 if (!isValidSyncEvent(event)) { 1940 return BAD_VALUE; 1941 } 1942 1943 Mutex::Autolock _l(mLock); 1944 1945 for (size_t i = 0; i < mTracks.size(); ++i) { 1946 sp<Track> track = mTracks[i]; 1947 if (event->triggerSession() == track->sessionId()) { 1948 track->setSyncEvent(event); 1949 return NO_ERROR; 1950 } 1951 } 1952 1953 return NAME_NOT_FOUND; 1954} 1955 1956bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 1957{ 1958 switch (event->type()) { 1959 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 1960 return true; 1961 default: 1962 break; 1963 } 1964 return false; 1965} 1966 1967// ---------------------------------------------------------------------------- 1968 1969AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1970 audio_io_handle_t id, uint32_t device, type_t type) 1971 : PlaybackThread(audioFlinger, output, id, device, type) 1972{ 1973 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1974 // FIXME - Current mixer implementation only supports stereo output 1975 if (mChannelCount == 1) { 1976 ALOGE("Invalid audio hardware channel count"); 1977 } 1978} 1979 1980AudioFlinger::MixerThread::~MixerThread() 1981{ 1982 delete mAudioMixer; 1983} 1984 1985class CpuStats { 1986public: 1987 CpuStats(); 1988 void sample(const String8 &title); 1989#ifdef DEBUG_CPU_USAGE 1990private: 1991 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 1992 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 1993 1994 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 1995 1996 int mCpuNum; // thread's current CPU number 1997 int mCpukHz; // frequency of thread's current CPU in kHz 1998#endif 1999}; 2000 2001CpuStats::CpuStats() 2002#ifdef DEBUG_CPU_USAGE 2003 : mCpuNum(-1), mCpukHz(-1) 2004#endif 2005{ 2006} 2007 2008void CpuStats::sample(const String8 &title) { 2009#ifdef DEBUG_CPU_USAGE 2010 // get current thread's delta CPU time in wall clock ns 2011 double wcNs; 2012 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2013 2014 // record sample for wall clock statistics 2015 if (valid) { 2016 mWcStats.sample(wcNs); 2017 } 2018 2019 // get the current CPU number 2020 int cpuNum = sched_getcpu(); 2021 2022 // get the current CPU frequency in kHz 2023 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2024 2025 // check if either CPU number or frequency changed 2026 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2027 mCpuNum = cpuNum; 2028 mCpukHz = cpukHz; 2029 // ignore sample for purposes of cycles 2030 valid = false; 2031 } 2032 2033 // if no change in CPU number or frequency, then record sample for cycle statistics 2034 if (valid && mCpukHz > 0) { 2035 double cycles = wcNs * cpukHz * 0.000001; 2036 mHzStats.sample(cycles); 2037 } 2038 2039 unsigned n = mWcStats.n(); 2040 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2041 if ((n & 127) == 1) { 2042 long long elapsed = mCpuUsage.elapsed(); 2043 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2044 double perLoop = elapsed / (double) n; 2045 double perLoop100 = perLoop * 0.01; 2046 double perLoop1k = perLoop * 0.001; 2047 double mean = mWcStats.mean(); 2048 double stddev = mWcStats.stddev(); 2049 double minimum = mWcStats.minimum(); 2050 double maximum = mWcStats.maximum(); 2051 double meanCycles = mHzStats.mean(); 2052 double stddevCycles = mHzStats.stddev(); 2053 double minCycles = mHzStats.minimum(); 2054 double maxCycles = mHzStats.maximum(); 2055 mCpuUsage.resetElapsed(); 2056 mWcStats.reset(); 2057 mHzStats.reset(); 2058 ALOGD("CPU usage for %s over past %.1f secs\n" 2059 " (%u mixer loops at %.1f mean ms per loop):\n" 2060 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2061 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2062 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2063 title.string(), 2064 elapsed * .000000001, n, perLoop * .000001, 2065 mean * .001, 2066 stddev * .001, 2067 minimum * .001, 2068 maximum * .001, 2069 mean / perLoop100, 2070 stddev / perLoop100, 2071 minimum / perLoop100, 2072 maximum / perLoop100, 2073 meanCycles / perLoop1k, 2074 stddevCycles / perLoop1k, 2075 minCycles / perLoop1k, 2076 maxCycles / perLoop1k); 2077 2078 } 2079 } 2080#endif 2081}; 2082 2083void AudioFlinger::PlaybackThread::checkSilentMode_l() 2084{ 2085 if (!mMasterMute) { 2086 char value[PROPERTY_VALUE_MAX]; 2087 if (property_get("ro.audio.silent", value, "0") > 0) { 2088 char *endptr; 2089 unsigned long ul = strtoul(value, &endptr, 0); 2090 if (*endptr == '\0' && ul != 0) { 2091 ALOGD("Silence is golden"); 2092 // The setprop command will not allow a property to be changed after 2093 // the first time it is set, so we don't have to worry about un-muting. 2094 setMasterMute_l(true); 2095 } 2096 } 2097 } 2098} 2099 2100bool AudioFlinger::PlaybackThread::threadLoop() 2101{ 2102 Vector< sp<Track> > tracksToRemove; 2103 2104 standbyTime = systemTime(); 2105 2106 // MIXER 2107 nsecs_t lastWarning = 0; 2108if (mType == MIXER) { 2109 longStandbyExit = false; 2110} 2111 2112 // DUPLICATING 2113 // FIXME could this be made local to while loop? 2114 writeFrames = 0; 2115 2116 cacheParameters_l(); 2117 sleepTime = idleSleepTime; 2118 2119if (mType == MIXER) { 2120 sleepTimeShift = 0; 2121} 2122 2123 CpuStats cpuStats; 2124 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2125 2126 acquireWakeLock(); 2127 2128 while (!exitPending()) 2129 { 2130 cpuStats.sample(myName); 2131 2132 Vector< sp<EffectChain> > effectChains; 2133 2134 processConfigEvents(); 2135 2136 { // scope for mLock 2137 2138 Mutex::Autolock _l(mLock); 2139 2140 if (checkForNewParameters_l()) { 2141 cacheParameters_l(); 2142 } 2143 2144 saveOutputTracks(); 2145 2146 // put audio hardware into standby after short delay 2147 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2148 mSuspended > 0)) { 2149 if (!mStandby) { 2150 2151 threadLoop_standby(); 2152 2153 mStandby = true; 2154 mBytesWritten = 0; 2155 } 2156 2157 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2158 // we're about to wait, flush the binder command buffer 2159 IPCThreadState::self()->flushCommands(); 2160 2161 clearOutputTracks(); 2162 2163 if (exitPending()) break; 2164 2165 releaseWakeLock_l(); 2166 // wait until we have something to do... 2167 ALOGV("%s going to sleep", myName.string()); 2168 mWaitWorkCV.wait(mLock); 2169 ALOGV("%s waking up", myName.string()); 2170 acquireWakeLock_l(); 2171 2172 mPrevMixerStatus = MIXER_IDLE; 2173 2174 checkSilentMode_l(); 2175 2176 standbyTime = systemTime() + standbyDelay; 2177 sleepTime = idleSleepTime; 2178 if (mType == MIXER) { 2179 sleepTimeShift = 0; 2180 } 2181 2182 continue; 2183 } 2184 } 2185 2186 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2187 // Shift in the new status; this could be a queue if it's 2188 // useful to filter the mixer status over several cycles. 2189 mPrevMixerStatus = mMixerStatus; 2190 mMixerStatus = newMixerStatus; 2191 2192 // prevent any changes in effect chain list and in each effect chain 2193 // during mixing and effect process as the audio buffers could be deleted 2194 // or modified if an effect is created or deleted 2195 lockEffectChains_l(effectChains); 2196 } 2197 2198 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2199 threadLoop_mix(); 2200 } else { 2201 threadLoop_sleepTime(); 2202 } 2203 2204 if (mSuspended > 0) { 2205 sleepTime = suspendSleepTimeUs(); 2206 } 2207 2208 // only process effects if we're going to write 2209 if (sleepTime == 0) { 2210 for (size_t i = 0; i < effectChains.size(); i ++) { 2211 effectChains[i]->process_l(); 2212 } 2213 } 2214 2215 // enable changes in effect chain 2216 unlockEffectChains(effectChains); 2217 2218 // sleepTime == 0 means we must write to audio hardware 2219 if (sleepTime == 0) { 2220 2221 threadLoop_write(); 2222 2223if (mType == MIXER) { 2224 // write blocked detection 2225 nsecs_t now = systemTime(); 2226 nsecs_t delta = now - mLastWriteTime; 2227 if (!mStandby && delta > maxPeriod) { 2228 mNumDelayedWrites++; 2229 if ((now - lastWarning) > kWarningThrottleNs) { 2230 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2231 ns2ms(delta), mNumDelayedWrites, this); 2232 lastWarning = now; 2233 } 2234 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2235 // a different threshold. Or completely removed for what it is worth anyway... 2236 if (mStandby) { 2237 longStandbyExit = true; 2238 } 2239 } 2240} 2241 2242 mStandby = false; 2243 } else { 2244 usleep(sleepTime); 2245 } 2246 2247 // finally let go of removed track(s), without the lock held 2248 // since we can't guarantee the destructors won't acquire that 2249 // same lock. 2250 tracksToRemove.clear(); 2251 2252 // FIXME I don't understand the need for this here; 2253 // it was in the original code but maybe the 2254 // assignment in saveOutputTracks() makes this unnecessary? 2255 clearOutputTracks(); 2256 2257 // Effect chains will be actually deleted here if they were removed from 2258 // mEffectChains list during mixing or effects processing 2259 effectChains.clear(); 2260 2261 // FIXME Note that the above .clear() is no longer necessary since effectChains 2262 // is now local to this block, but will keep it for now (at least until merge done). 2263 } 2264 2265if (mType == MIXER || mType == DIRECT) { 2266 // put output stream into standby mode 2267 if (!mStandby) { 2268 mOutput->stream->common.standby(&mOutput->stream->common); 2269 } 2270} 2271if (mType == DUPLICATING) { 2272 // for DuplicatingThread, standby mode is handled by the outputTracks 2273} 2274 2275 releaseWakeLock(); 2276 2277 ALOGV("Thread %p type %d exiting", this, mType); 2278 return false; 2279} 2280 2281// shared by MIXER and DIRECT, overridden by DUPLICATING 2282void AudioFlinger::PlaybackThread::threadLoop_write() 2283{ 2284 // FIXME rewrite to reduce number of system calls 2285 mLastWriteTime = systemTime(); 2286 mInWrite = true; 2287 mBytesWritten += mixBufferSize; 2288 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2289 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2290 mNumWrites++; 2291 mInWrite = false; 2292} 2293 2294// shared by MIXER and DIRECT, overridden by DUPLICATING 2295void AudioFlinger::PlaybackThread::threadLoop_standby() 2296{ 2297 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2298 mOutput->stream->common.standby(&mOutput->stream->common); 2299} 2300 2301void AudioFlinger::MixerThread::threadLoop_mix() 2302{ 2303 // obtain the presentation timestamp of the next output buffer 2304 int64_t pts; 2305 status_t status = INVALID_OPERATION; 2306 2307 if (NULL != mOutput->stream->get_next_write_timestamp) { 2308 status = mOutput->stream->get_next_write_timestamp( 2309 mOutput->stream, &pts); 2310 } 2311 2312 if (status != NO_ERROR) { 2313 pts = AudioBufferProvider::kInvalidPTS; 2314 } 2315 2316 // mix buffers... 2317 mAudioMixer->process(pts); 2318 // increase sleep time progressively when application underrun condition clears. 2319 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2320 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2321 // such that we would underrun the audio HAL. 2322 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2323 sleepTimeShift--; 2324 } 2325 sleepTime = 0; 2326 standbyTime = systemTime() + standbyDelay; 2327 //TODO: delay standby when effects have a tail 2328} 2329 2330void AudioFlinger::MixerThread::threadLoop_sleepTime() 2331{ 2332 // If no tracks are ready, sleep once for the duration of an output 2333 // buffer size, then write 0s to the output 2334 if (sleepTime == 0) { 2335 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2336 sleepTime = activeSleepTime >> sleepTimeShift; 2337 if (sleepTime < kMinThreadSleepTimeUs) { 2338 sleepTime = kMinThreadSleepTimeUs; 2339 } 2340 // reduce sleep time in case of consecutive application underruns to avoid 2341 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2342 // duration we would end up writing less data than needed by the audio HAL if 2343 // the condition persists. 2344 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2345 sleepTimeShift++; 2346 } 2347 } else { 2348 sleepTime = idleSleepTime; 2349 } 2350 } else if (mBytesWritten != 0 || 2351 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2352 memset (mMixBuffer, 0, mixBufferSize); 2353 sleepTime = 0; 2354 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2355 } 2356 // TODO add standby time extension fct of effect tail 2357} 2358 2359// prepareTracks_l() must be called with ThreadBase::mLock held 2360AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2361 Vector< sp<Track> > *tracksToRemove) 2362{ 2363 2364 mixer_state mixerStatus = MIXER_IDLE; 2365 // find out which tracks need to be processed 2366 size_t count = mActiveTracks.size(); 2367 size_t mixedTracks = 0; 2368 size_t tracksWithEffect = 0; 2369 2370 float masterVolume = mMasterVolume; 2371 bool masterMute = mMasterMute; 2372 2373 if (masterMute) { 2374 masterVolume = 0; 2375 } 2376 // Delegate master volume control to effect in output mix effect chain if needed 2377 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2378 if (chain != 0) { 2379 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2380 chain->setVolume_l(&v, &v); 2381 masterVolume = (float)((v + (1 << 23)) >> 24); 2382 chain.clear(); 2383 } 2384 2385 for (size_t i=0 ; i<count ; i++) { 2386 sp<Track> t = mActiveTracks[i].promote(); 2387 if (t == 0) continue; 2388 2389 // this const just means the local variable doesn't change 2390 Track* const track = t.get(); 2391 audio_track_cblk_t* cblk = track->cblk(); 2392 2393 // The first time a track is added we wait 2394 // for all its buffers to be filled before processing it 2395 int name = track->name(); 2396 // make sure that we have enough frames to mix one full buffer. 2397 // enforce this condition only once to enable draining the buffer in case the client 2398 // app does not call stop() and relies on underrun to stop: 2399 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2400 // during last round 2401 uint32_t minFrames = 1; 2402 if (!track->isStopped() && !track->isPausing() && 2403 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2404 if (t->sampleRate() == (int)mSampleRate) { 2405 minFrames = mFrameCount; 2406 } else { 2407 // +1 for rounding and +1 for additional sample needed for interpolation 2408 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2409 // add frames already consumed but not yet released by the resampler 2410 // because cblk->framesReady() will include these frames 2411 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2412 // the minimum track buffer size is normally twice the number of frames necessary 2413 // to fill one buffer and the resampler should not leave more than one buffer worth 2414 // of unreleased frames after each pass, but just in case... 2415 ALOG_ASSERT(minFrames <= cblk->frameCount); 2416 } 2417 } 2418 if ((track->framesReady() >= minFrames) && track->isReady() && 2419 !track->isPaused() && !track->isTerminated()) 2420 { 2421 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2422 2423 mixedTracks++; 2424 2425 // track->mainBuffer() != mMixBuffer means there is an effect chain 2426 // connected to the track 2427 chain.clear(); 2428 if (track->mainBuffer() != mMixBuffer) { 2429 chain = getEffectChain_l(track->sessionId()); 2430 // Delegate volume control to effect in track effect chain if needed 2431 if (chain != 0) { 2432 tracksWithEffect++; 2433 } else { 2434 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2435 name, track->sessionId()); 2436 } 2437 } 2438 2439 2440 int param = AudioMixer::VOLUME; 2441 if (track->mFillingUpStatus == Track::FS_FILLED) { 2442 // no ramp for the first volume setting 2443 track->mFillingUpStatus = Track::FS_ACTIVE; 2444 if (track->mState == TrackBase::RESUMING) { 2445 track->mState = TrackBase::ACTIVE; 2446 param = AudioMixer::RAMP_VOLUME; 2447 } 2448 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2449 } else if (cblk->server != 0) { 2450 // If the track is stopped before the first frame was mixed, 2451 // do not apply ramp 2452 param = AudioMixer::RAMP_VOLUME; 2453 } 2454 2455 // compute volume for this track 2456 uint32_t vl, vr, va; 2457 if (track->isMuted() || track->isPausing() || 2458 mStreamTypes[track->streamType()].mute) { 2459 vl = vr = va = 0; 2460 if (track->isPausing()) { 2461 track->setPaused(); 2462 } 2463 } else { 2464 2465 // read original volumes with volume control 2466 float typeVolume = mStreamTypes[track->streamType()].volume; 2467 float v = masterVolume * typeVolume; 2468 uint32_t vlr = cblk->getVolumeLR(); 2469 vl = vlr & 0xFFFF; 2470 vr = vlr >> 16; 2471 // track volumes come from shared memory, so can't be trusted and must be clamped 2472 if (vl > MAX_GAIN_INT) { 2473 ALOGV("Track left volume out of range: %04X", vl); 2474 vl = MAX_GAIN_INT; 2475 } 2476 if (vr > MAX_GAIN_INT) { 2477 ALOGV("Track right volume out of range: %04X", vr); 2478 vr = MAX_GAIN_INT; 2479 } 2480 // now apply the master volume and stream type volume 2481 vl = (uint32_t)(v * vl) << 12; 2482 vr = (uint32_t)(v * vr) << 12; 2483 // assuming master volume and stream type volume each go up to 1.0, 2484 // vl and vr are now in 8.24 format 2485 2486 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2487 // send level comes from shared memory and so may be corrupt 2488 if (sendLevel > MAX_GAIN_INT) { 2489 ALOGV("Track send level out of range: %04X", sendLevel); 2490 sendLevel = MAX_GAIN_INT; 2491 } 2492 va = (uint32_t)(v * sendLevel); 2493 } 2494 // Delegate volume control to effect in track effect chain if needed 2495 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2496 // Do not ramp volume if volume is controlled by effect 2497 param = AudioMixer::VOLUME; 2498 track->mHasVolumeController = true; 2499 } else { 2500 // force no volume ramp when volume controller was just disabled or removed 2501 // from effect chain to avoid volume spike 2502 if (track->mHasVolumeController) { 2503 param = AudioMixer::VOLUME; 2504 } 2505 track->mHasVolumeController = false; 2506 } 2507 2508 // Convert volumes from 8.24 to 4.12 format 2509 // This additional clamping is needed in case chain->setVolume_l() overshot 2510 vl = (vl + (1 << 11)) >> 12; 2511 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2512 vr = (vr + (1 << 11)) >> 12; 2513 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2514 2515 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2516 2517 // XXX: these things DON'T need to be done each time 2518 mAudioMixer->setBufferProvider(name, track); 2519 mAudioMixer->enable(name); 2520 2521 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2522 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2523 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2524 mAudioMixer->setParameter( 2525 name, 2526 AudioMixer::TRACK, 2527 AudioMixer::FORMAT, (void *)track->format()); 2528 mAudioMixer->setParameter( 2529 name, 2530 AudioMixer::TRACK, 2531 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2532 mAudioMixer->setParameter( 2533 name, 2534 AudioMixer::RESAMPLE, 2535 AudioMixer::SAMPLE_RATE, 2536 (void *)(cblk->sampleRate)); 2537 mAudioMixer->setParameter( 2538 name, 2539 AudioMixer::TRACK, 2540 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2541 mAudioMixer->setParameter( 2542 name, 2543 AudioMixer::TRACK, 2544 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2545 2546 // reset retry count 2547 track->mRetryCount = kMaxTrackRetries; 2548 2549 // If one track is ready, set the mixer ready if: 2550 // - the mixer was not ready during previous round OR 2551 // - no other track is not ready 2552 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2553 mixerStatus != MIXER_TRACKS_ENABLED) { 2554 mixerStatus = MIXER_TRACKS_READY; 2555 } 2556 } else { 2557 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2558 if (track->isStopped()) { 2559 track->reset(); 2560 } 2561 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2562 // We have consumed all the buffers of this track. 2563 // Remove it from the list of active tracks. 2564 // TODO: use actual buffer filling status instead of latency when available from 2565 // audio HAL 2566 size_t audioHALFrames = 2567 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2568 size_t framesWritten = 2569 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2570 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2571 tracksToRemove->add(track); 2572 } 2573 } else { 2574 // No buffers for this track. Give it a few chances to 2575 // fill a buffer, then remove it from active list. 2576 if (--(track->mRetryCount) <= 0) { 2577 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2578 tracksToRemove->add(track); 2579 // indicate to client process that the track was disabled because of underrun 2580 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2581 // If one track is not ready, mark the mixer also not ready if: 2582 // - the mixer was ready during previous round OR 2583 // - no other track is ready 2584 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2585 mixerStatus != MIXER_TRACKS_READY) { 2586 mixerStatus = MIXER_TRACKS_ENABLED; 2587 } 2588 } 2589 mAudioMixer->disable(name); 2590 } 2591 } 2592 2593 // remove all the tracks that need to be... 2594 count = tracksToRemove->size(); 2595 if (CC_UNLIKELY(count)) { 2596 for (size_t i=0 ; i<count ; i++) { 2597 const sp<Track>& track = tracksToRemove->itemAt(i); 2598 mActiveTracks.remove(track); 2599 if (track->mainBuffer() != mMixBuffer) { 2600 chain = getEffectChain_l(track->sessionId()); 2601 if (chain != 0) { 2602 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2603 chain->decActiveTrackCnt(); 2604 } 2605 } 2606 if (track->isTerminated()) { 2607 removeTrack_l(track); 2608 } 2609 } 2610 } 2611 2612 // mix buffer must be cleared if all tracks are connected to an 2613 // effect chain as in this case the mixer will not write to 2614 // mix buffer and track effects will accumulate into it 2615 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2616 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2617 } 2618 2619 return mixerStatus; 2620} 2621 2622/* 2623The derived values that are cached: 2624 - mixBufferSize from frame count * frame size 2625 - activeSleepTime from activeSleepTimeUs() 2626 - idleSleepTime from idleSleepTimeUs() 2627 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2628 - maxPeriod from frame count and sample rate (MIXER only) 2629 2630The parameters that affect these derived values are: 2631 - frame count 2632 - frame size 2633 - sample rate 2634 - device type: A2DP or not 2635 - device latency 2636 - format: PCM or not 2637 - active sleep time 2638 - idle sleep time 2639*/ 2640 2641void AudioFlinger::PlaybackThread::cacheParameters_l() 2642{ 2643 mixBufferSize = mFrameCount * mFrameSize; 2644 activeSleepTime = activeSleepTimeUs(); 2645 idleSleepTime = idleSleepTimeUs(); 2646} 2647 2648void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2649{ 2650 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2651 this, streamType, mTracks.size()); 2652 Mutex::Autolock _l(mLock); 2653 2654 size_t size = mTracks.size(); 2655 for (size_t i = 0; i < size; i++) { 2656 sp<Track> t = mTracks[i]; 2657 if (t->streamType() == streamType) { 2658 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2659 t->mCblk->cv.signal(); 2660 } 2661 } 2662} 2663 2664// getTrackName_l() must be called with ThreadBase::mLock held 2665int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 2666{ 2667 int name = mAudioMixer->getTrackName(); 2668 if (name >= 0) { 2669 mAudioMixer->setParameter(name, 2670 AudioMixer::TRACK, 2671 AudioMixer::CHANNEL_MASK, (void *)channelMask); 2672 } 2673 return name; 2674} 2675 2676// deleteTrackName_l() must be called with ThreadBase::mLock held 2677void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2678{ 2679 ALOGV("remove track (%d) and delete from mixer", name); 2680 mAudioMixer->deleteTrackName(name); 2681} 2682 2683// checkForNewParameters_l() must be called with ThreadBase::mLock held 2684bool AudioFlinger::MixerThread::checkForNewParameters_l() 2685{ 2686 bool reconfig = false; 2687 2688 while (!mNewParameters.isEmpty()) { 2689 status_t status = NO_ERROR; 2690 String8 keyValuePair = mNewParameters[0]; 2691 AudioParameter param = AudioParameter(keyValuePair); 2692 int value; 2693 2694 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2695 reconfig = true; 2696 } 2697 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2698 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2699 status = BAD_VALUE; 2700 } else { 2701 reconfig = true; 2702 } 2703 } 2704 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2705 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2706 status = BAD_VALUE; 2707 } else { 2708 reconfig = true; 2709 } 2710 } 2711 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2712 // do not accept frame count changes if tracks are open as the track buffer 2713 // size depends on frame count and correct behavior would not be guaranteed 2714 // if frame count is changed after track creation 2715 if (!mTracks.isEmpty()) { 2716 status = INVALID_OPERATION; 2717 } else { 2718 reconfig = true; 2719 } 2720 } 2721 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2722#ifdef ADD_BATTERY_DATA 2723 // when changing the audio output device, call addBatteryData to notify 2724 // the change 2725 if ((int)mDevice != value) { 2726 uint32_t params = 0; 2727 // check whether speaker is on 2728 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2729 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2730 } 2731 2732 int deviceWithoutSpeaker 2733 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2734 // check if any other device (except speaker) is on 2735 if (value & deviceWithoutSpeaker ) { 2736 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2737 } 2738 2739 if (params != 0) { 2740 addBatteryData(params); 2741 } 2742 } 2743#endif 2744 2745 // forward device change to effects that have requested to be 2746 // aware of attached audio device. 2747 mDevice = (uint32_t)value; 2748 for (size_t i = 0; i < mEffectChains.size(); i++) { 2749 mEffectChains[i]->setDevice_l(mDevice); 2750 } 2751 } 2752 2753 if (status == NO_ERROR) { 2754 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2755 keyValuePair.string()); 2756 if (!mStandby && status == INVALID_OPERATION) { 2757 mOutput->stream->common.standby(&mOutput->stream->common); 2758 mStandby = true; 2759 mBytesWritten = 0; 2760 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2761 keyValuePair.string()); 2762 } 2763 if (status == NO_ERROR && reconfig) { 2764 delete mAudioMixer; 2765 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2766 mAudioMixer = NULL; 2767 readOutputParameters(); 2768 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2769 for (size_t i = 0; i < mTracks.size() ; i++) { 2770 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 2771 if (name < 0) break; 2772 mTracks[i]->mName = name; 2773 // limit track sample rate to 2 x new output sample rate 2774 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2775 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2776 } 2777 } 2778 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2779 } 2780 } 2781 2782 mNewParameters.removeAt(0); 2783 2784 mParamStatus = status; 2785 mParamCond.signal(); 2786 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2787 // already timed out waiting for the status and will never signal the condition. 2788 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2789 } 2790 return reconfig; 2791} 2792 2793status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2794{ 2795 const size_t SIZE = 256; 2796 char buffer[SIZE]; 2797 String8 result; 2798 2799 PlaybackThread::dumpInternals(fd, args); 2800 2801 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2802 result.append(buffer); 2803 write(fd, result.string(), result.size()); 2804 return NO_ERROR; 2805} 2806 2807uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 2808{ 2809 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2810} 2811 2812uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 2813{ 2814 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2815} 2816 2817void AudioFlinger::MixerThread::cacheParameters_l() 2818{ 2819 PlaybackThread::cacheParameters_l(); 2820 2821 // FIXME: Relaxed timing because of a certain device that can't meet latency 2822 // Should be reduced to 2x after the vendor fixes the driver issue 2823 // increase threshold again due to low power audio mode. The way this warning 2824 // threshold is calculated and its usefulness should be reconsidered anyway. 2825 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2826} 2827 2828// ---------------------------------------------------------------------------- 2829AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2830 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2831 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2832 // mLeftVolFloat, mRightVolFloat 2833 // mLeftVolShort, mRightVolShort 2834{ 2835} 2836 2837AudioFlinger::DirectOutputThread::~DirectOutputThread() 2838{ 2839} 2840 2841AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2842 Vector< sp<Track> > *tracksToRemove 2843) 2844{ 2845 sp<Track> trackToRemove; 2846 2847 mixer_state mixerStatus = MIXER_IDLE; 2848 2849 // find out which tracks need to be processed 2850 if (mActiveTracks.size() != 0) { 2851 sp<Track> t = mActiveTracks[0].promote(); 2852 // The track died recently 2853 if (t == 0) return MIXER_IDLE; 2854 2855 Track* const track = t.get(); 2856 audio_track_cblk_t* cblk = track->cblk(); 2857 2858 // The first time a track is added we wait 2859 // for all its buffers to be filled before processing it 2860 if (cblk->framesReady() && track->isReady() && 2861 !track->isPaused() && !track->isTerminated()) 2862 { 2863 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2864 2865 if (track->mFillingUpStatus == Track::FS_FILLED) { 2866 track->mFillingUpStatus = Track::FS_ACTIVE; 2867 mLeftVolFloat = mRightVolFloat = 0; 2868 mLeftVolShort = mRightVolShort = 0; 2869 if (track->mState == TrackBase::RESUMING) { 2870 track->mState = TrackBase::ACTIVE; 2871 rampVolume = true; 2872 } 2873 } else if (cblk->server != 0) { 2874 // If the track is stopped before the first frame was mixed, 2875 // do not apply ramp 2876 rampVolume = true; 2877 } 2878 // compute volume for this track 2879 float left, right; 2880 if (track->isMuted() || mMasterMute || track->isPausing() || 2881 mStreamTypes[track->streamType()].mute) { 2882 left = right = 0; 2883 if (track->isPausing()) { 2884 track->setPaused(); 2885 } 2886 } else { 2887 float typeVolume = mStreamTypes[track->streamType()].volume; 2888 float v = mMasterVolume * typeVolume; 2889 uint32_t vlr = cblk->getVolumeLR(); 2890 float v_clamped = v * (vlr & 0xFFFF); 2891 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2892 left = v_clamped/MAX_GAIN; 2893 v_clamped = v * (vlr >> 16); 2894 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2895 right = v_clamped/MAX_GAIN; 2896 } 2897 2898 if (left != mLeftVolFloat || right != mRightVolFloat) { 2899 mLeftVolFloat = left; 2900 mRightVolFloat = right; 2901 2902 // If audio HAL implements volume control, 2903 // force software volume to nominal value 2904 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2905 left = 1.0f; 2906 right = 1.0f; 2907 } 2908 2909 // Convert volumes from float to 8.24 2910 uint32_t vl = (uint32_t)(left * (1 << 24)); 2911 uint32_t vr = (uint32_t)(right * (1 << 24)); 2912 2913 // Delegate volume control to effect in track effect chain if needed 2914 // only one effect chain can be present on DirectOutputThread, so if 2915 // there is one, the track is connected to it 2916 if (!mEffectChains.isEmpty()) { 2917 // Do not ramp volume if volume is controlled by effect 2918 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2919 rampVolume = false; 2920 } 2921 } 2922 2923 // Convert volumes from 8.24 to 4.12 format 2924 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2925 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2926 leftVol = (uint16_t)v_clamped; 2927 v_clamped = (vr + (1 << 11)) >> 12; 2928 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2929 rightVol = (uint16_t)v_clamped; 2930 } else { 2931 leftVol = mLeftVolShort; 2932 rightVol = mRightVolShort; 2933 rampVolume = false; 2934 } 2935 2936 // reset retry count 2937 track->mRetryCount = kMaxTrackRetriesDirect; 2938 mActiveTrack = t; 2939 mixerStatus = MIXER_TRACKS_READY; 2940 } else { 2941 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2942 if (track->isStopped()) { 2943 track->reset(); 2944 } 2945 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2946 // We have consumed all the buffers of this track. 2947 // Remove it from the list of active tracks. 2948 // TODO: implement behavior for compressed audio 2949 size_t audioHALFrames = 2950 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2951 size_t framesWritten = 2952 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2953 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2954 trackToRemove = track; 2955 } 2956 } else { 2957 // No buffers for this track. Give it a few chances to 2958 // fill a buffer, then remove it from active list. 2959 if (--(track->mRetryCount) <= 0) { 2960 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2961 trackToRemove = track; 2962 } else { 2963 mixerStatus = MIXER_TRACKS_ENABLED; 2964 } 2965 } 2966 } 2967 } 2968 2969 // FIXME merge this with similar code for removing multiple tracks 2970 // remove all the tracks that need to be... 2971 if (CC_UNLIKELY(trackToRemove != 0)) { 2972 tracksToRemove->add(trackToRemove); 2973 mActiveTracks.remove(trackToRemove); 2974 if (!mEffectChains.isEmpty()) { 2975 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2976 trackToRemove->sessionId()); 2977 mEffectChains[0]->decActiveTrackCnt(); 2978 } 2979 if (trackToRemove->isTerminated()) { 2980 removeTrack_l(trackToRemove); 2981 } 2982 } 2983 2984 return mixerStatus; 2985} 2986 2987void AudioFlinger::DirectOutputThread::threadLoop_mix() 2988{ 2989 AudioBufferProvider::Buffer buffer; 2990 size_t frameCount = mFrameCount; 2991 int8_t *curBuf = (int8_t *)mMixBuffer; 2992 // output audio to hardware 2993 while (frameCount) { 2994 buffer.frameCount = frameCount; 2995 mActiveTrack->getNextBuffer(&buffer); 2996 if (CC_UNLIKELY(buffer.raw == NULL)) { 2997 memset(curBuf, 0, frameCount * mFrameSize); 2998 break; 2999 } 3000 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3001 frameCount -= buffer.frameCount; 3002 curBuf += buffer.frameCount * mFrameSize; 3003 mActiveTrack->releaseBuffer(&buffer); 3004 } 3005 sleepTime = 0; 3006 standbyTime = systemTime() + standbyDelay; 3007 mActiveTrack.clear(); 3008 3009 // apply volume 3010 3011 // Do not apply volume on compressed audio 3012 if (!audio_is_linear_pcm(mFormat)) { 3013 return; 3014 } 3015 3016 // convert to signed 16 bit before volume calculation 3017 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3018 size_t count = mFrameCount * mChannelCount; 3019 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3020 int16_t *dst = mMixBuffer + count-1; 3021 while (count--) { 3022 *dst-- = (int16_t)(*src--^0x80) << 8; 3023 } 3024 } 3025 3026 frameCount = mFrameCount; 3027 int16_t *out = mMixBuffer; 3028 if (rampVolume) { 3029 if (mChannelCount == 1) { 3030 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3031 int32_t vlInc = d / (int32_t)frameCount; 3032 int32_t vl = ((int32_t)mLeftVolShort << 16); 3033 do { 3034 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3035 out++; 3036 vl += vlInc; 3037 } while (--frameCount); 3038 3039 } else { 3040 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3041 int32_t vlInc = d / (int32_t)frameCount; 3042 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3043 int32_t vrInc = d / (int32_t)frameCount; 3044 int32_t vl = ((int32_t)mLeftVolShort << 16); 3045 int32_t vr = ((int32_t)mRightVolShort << 16); 3046 do { 3047 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3048 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3049 out += 2; 3050 vl += vlInc; 3051 vr += vrInc; 3052 } while (--frameCount); 3053 } 3054 } else { 3055 if (mChannelCount == 1) { 3056 do { 3057 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3058 out++; 3059 } while (--frameCount); 3060 } else { 3061 do { 3062 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3063 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3064 out += 2; 3065 } while (--frameCount); 3066 } 3067 } 3068 3069 // convert back to unsigned 8 bit after volume calculation 3070 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3071 size_t count = mFrameCount * mChannelCount; 3072 int16_t *src = mMixBuffer; 3073 uint8_t *dst = (uint8_t *)mMixBuffer; 3074 while (count--) { 3075 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3076 } 3077 } 3078 3079 mLeftVolShort = leftVol; 3080 mRightVolShort = rightVol; 3081} 3082 3083void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3084{ 3085 if (sleepTime == 0) { 3086 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3087 sleepTime = activeSleepTime; 3088 } else { 3089 sleepTime = idleSleepTime; 3090 } 3091 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3092 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3093 sleepTime = 0; 3094 } 3095} 3096 3097// getTrackName_l() must be called with ThreadBase::mLock held 3098int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3099{ 3100 return 0; 3101} 3102 3103// deleteTrackName_l() must be called with ThreadBase::mLock held 3104void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3105{ 3106} 3107 3108// checkForNewParameters_l() must be called with ThreadBase::mLock held 3109bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3110{ 3111 bool reconfig = false; 3112 3113 while (!mNewParameters.isEmpty()) { 3114 status_t status = NO_ERROR; 3115 String8 keyValuePair = mNewParameters[0]; 3116 AudioParameter param = AudioParameter(keyValuePair); 3117 int value; 3118 3119 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3120 // do not accept frame count changes if tracks are open as the track buffer 3121 // size depends on frame count and correct behavior would not be garantied 3122 // if frame count is changed after track creation 3123 if (!mTracks.isEmpty()) { 3124 status = INVALID_OPERATION; 3125 } else { 3126 reconfig = true; 3127 } 3128 } 3129 if (status == NO_ERROR) { 3130 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3131 keyValuePair.string()); 3132 if (!mStandby && status == INVALID_OPERATION) { 3133 mOutput->stream->common.standby(&mOutput->stream->common); 3134 mStandby = true; 3135 mBytesWritten = 0; 3136 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3137 keyValuePair.string()); 3138 } 3139 if (status == NO_ERROR && reconfig) { 3140 readOutputParameters(); 3141 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3142 } 3143 } 3144 3145 mNewParameters.removeAt(0); 3146 3147 mParamStatus = status; 3148 mParamCond.signal(); 3149 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3150 // already timed out waiting for the status and will never signal the condition. 3151 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3152 } 3153 return reconfig; 3154} 3155 3156uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3157{ 3158 uint32_t time; 3159 if (audio_is_linear_pcm(mFormat)) { 3160 time = PlaybackThread::activeSleepTimeUs(); 3161 } else { 3162 time = 10000; 3163 } 3164 return time; 3165} 3166 3167uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3168{ 3169 uint32_t time; 3170 if (audio_is_linear_pcm(mFormat)) { 3171 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3172 } else { 3173 time = 10000; 3174 } 3175 return time; 3176} 3177 3178uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3179{ 3180 uint32_t time; 3181 if (audio_is_linear_pcm(mFormat)) { 3182 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3183 } else { 3184 time = 10000; 3185 } 3186 return time; 3187} 3188 3189void AudioFlinger::DirectOutputThread::cacheParameters_l() 3190{ 3191 PlaybackThread::cacheParameters_l(); 3192 3193 // use shorter standby delay as on normal output to release 3194 // hardware resources as soon as possible 3195 standbyDelay = microseconds(activeSleepTime*2); 3196} 3197 3198// ---------------------------------------------------------------------------- 3199 3200AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3201 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3202 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3203 mWaitTimeMs(UINT_MAX) 3204{ 3205 addOutputTrack(mainThread); 3206} 3207 3208AudioFlinger::DuplicatingThread::~DuplicatingThread() 3209{ 3210 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3211 mOutputTracks[i]->destroy(); 3212 } 3213} 3214 3215void AudioFlinger::DuplicatingThread::threadLoop_mix() 3216{ 3217 // mix buffers... 3218 if (outputsReady(outputTracks)) { 3219 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3220 } else { 3221 memset(mMixBuffer, 0, mixBufferSize); 3222 } 3223 sleepTime = 0; 3224 writeFrames = mFrameCount; 3225} 3226 3227void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3228{ 3229 if (sleepTime == 0) { 3230 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3231 sleepTime = activeSleepTime; 3232 } else { 3233 sleepTime = idleSleepTime; 3234 } 3235 } else if (mBytesWritten != 0) { 3236 // flush remaining overflow buffers in output tracks 3237 for (size_t i = 0; i < outputTracks.size(); i++) { 3238 if (outputTracks[i]->isActive()) { 3239 sleepTime = 0; 3240 writeFrames = 0; 3241 memset(mMixBuffer, 0, mixBufferSize); 3242 break; 3243 } 3244 } 3245 } 3246} 3247 3248void AudioFlinger::DuplicatingThread::threadLoop_write() 3249{ 3250 standbyTime = systemTime() + standbyDelay; 3251 for (size_t i = 0; i < outputTracks.size(); i++) { 3252 outputTracks[i]->write(mMixBuffer, writeFrames); 3253 } 3254 mBytesWritten += mixBufferSize; 3255} 3256 3257void AudioFlinger::DuplicatingThread::threadLoop_standby() 3258{ 3259 // DuplicatingThread implements standby by stopping all tracks 3260 for (size_t i = 0; i < outputTracks.size(); i++) { 3261 outputTracks[i]->stop(); 3262 } 3263} 3264 3265void AudioFlinger::DuplicatingThread::saveOutputTracks() 3266{ 3267 outputTracks = mOutputTracks; 3268} 3269 3270void AudioFlinger::DuplicatingThread::clearOutputTracks() 3271{ 3272 outputTracks.clear(); 3273} 3274 3275void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3276{ 3277 Mutex::Autolock _l(mLock); 3278 // FIXME explain this formula 3279 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3280 OutputTrack *outputTrack = new OutputTrack(thread, 3281 this, 3282 mSampleRate, 3283 mFormat, 3284 mChannelMask, 3285 frameCount); 3286 if (outputTrack->cblk() != NULL) { 3287 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3288 mOutputTracks.add(outputTrack); 3289 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3290 updateWaitTime_l(); 3291 } 3292} 3293 3294void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3295{ 3296 Mutex::Autolock _l(mLock); 3297 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3298 if (mOutputTracks[i]->thread() == thread) { 3299 mOutputTracks[i]->destroy(); 3300 mOutputTracks.removeAt(i); 3301 updateWaitTime_l(); 3302 return; 3303 } 3304 } 3305 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3306} 3307 3308// caller must hold mLock 3309void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3310{ 3311 mWaitTimeMs = UINT_MAX; 3312 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3313 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3314 if (strong != 0) { 3315 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3316 if (waitTimeMs < mWaitTimeMs) { 3317 mWaitTimeMs = waitTimeMs; 3318 } 3319 } 3320 } 3321} 3322 3323 3324bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3325{ 3326 for (size_t i = 0; i < outputTracks.size(); i++) { 3327 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3328 if (thread == 0) { 3329 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3330 return false; 3331 } 3332 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3333 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3334 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3335 return false; 3336 } 3337 } 3338 return true; 3339} 3340 3341uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3342{ 3343 return (mWaitTimeMs * 1000) / 2; 3344} 3345 3346void AudioFlinger::DuplicatingThread::cacheParameters_l() 3347{ 3348 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3349 updateWaitTime_l(); 3350 3351 MixerThread::cacheParameters_l(); 3352} 3353 3354// ---------------------------------------------------------------------------- 3355 3356// TrackBase constructor must be called with AudioFlinger::mLock held 3357AudioFlinger::ThreadBase::TrackBase::TrackBase( 3358 ThreadBase *thread, 3359 const sp<Client>& client, 3360 uint32_t sampleRate, 3361 audio_format_t format, 3362 uint32_t channelMask, 3363 int frameCount, 3364 const sp<IMemory>& sharedBuffer, 3365 int sessionId) 3366 : RefBase(), 3367 mThread(thread), 3368 mClient(client), 3369 mCblk(NULL), 3370 // mBuffer 3371 // mBufferEnd 3372 mFrameCount(0), 3373 mState(IDLE), 3374 mFormat(format), 3375 mStepServerFailed(false), 3376 mSessionId(sessionId) 3377 // mChannelCount 3378 // mChannelMask 3379{ 3380 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3381 3382 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3383 size_t size = sizeof(audio_track_cblk_t); 3384 uint8_t channelCount = popcount(channelMask); 3385 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3386 if (sharedBuffer == 0) { 3387 size += bufferSize; 3388 } 3389 3390 if (client != NULL) { 3391 mCblkMemory = client->heap()->allocate(size); 3392 if (mCblkMemory != 0) { 3393 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3394 if (mCblk != NULL) { // construct the shared structure in-place. 3395 new(mCblk) audio_track_cblk_t(); 3396 // clear all buffers 3397 mCblk->frameCount = frameCount; 3398 mCblk->sampleRate = sampleRate; 3399// uncomment the following lines to quickly test 32-bit wraparound 3400// mCblk->user = 0xffff0000; 3401// mCblk->server = 0xffff0000; 3402// mCblk->userBase = 0xffff0000; 3403// mCblk->serverBase = 0xffff0000; 3404 mChannelCount = channelCount; 3405 mChannelMask = channelMask; 3406 if (sharedBuffer == 0) { 3407 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3408 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3409 // Force underrun condition to avoid false underrun callback until first data is 3410 // written to buffer (other flags are cleared) 3411 mCblk->flags = CBLK_UNDERRUN_ON; 3412 } else { 3413 mBuffer = sharedBuffer->pointer(); 3414 } 3415 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3416 } 3417 } else { 3418 ALOGE("not enough memory for AudioTrack size=%u", size); 3419 client->heap()->dump("AudioTrack"); 3420 return; 3421 } 3422 } else { 3423 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3424 // construct the shared structure in-place. 3425 new(mCblk) audio_track_cblk_t(); 3426 // clear all buffers 3427 mCblk->frameCount = frameCount; 3428 mCblk->sampleRate = sampleRate; 3429// uncomment the following lines to quickly test 32-bit wraparound 3430// mCblk->user = 0xffff0000; 3431// mCblk->server = 0xffff0000; 3432// mCblk->userBase = 0xffff0000; 3433// mCblk->serverBase = 0xffff0000; 3434 mChannelCount = channelCount; 3435 mChannelMask = channelMask; 3436 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3437 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3438 // Force underrun condition to avoid false underrun callback until first data is 3439 // written to buffer (other flags are cleared) 3440 mCblk->flags = CBLK_UNDERRUN_ON; 3441 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3442 } 3443} 3444 3445AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3446{ 3447 if (mCblk != NULL) { 3448 if (mClient == 0) { 3449 delete mCblk; 3450 } else { 3451 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3452 } 3453 } 3454 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3455 if (mClient != 0) { 3456 // Client destructor must run with AudioFlinger mutex locked 3457 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3458 // If the client's reference count drops to zero, the associated destructor 3459 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3460 // relying on the automatic clear() at end of scope. 3461 mClient.clear(); 3462 } 3463} 3464 3465// AudioBufferProvider interface 3466// getNextBuffer() = 0; 3467// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3468void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3469{ 3470 buffer->raw = NULL; 3471 mFrameCount = buffer->frameCount; 3472 (void) step(); // ignore return value of step() 3473 buffer->frameCount = 0; 3474} 3475 3476bool AudioFlinger::ThreadBase::TrackBase::step() { 3477 bool result; 3478 audio_track_cblk_t* cblk = this->cblk(); 3479 3480 result = cblk->stepServer(mFrameCount); 3481 if (!result) { 3482 ALOGV("stepServer failed acquiring cblk mutex"); 3483 mStepServerFailed = true; 3484 } 3485 return result; 3486} 3487 3488void AudioFlinger::ThreadBase::TrackBase::reset() { 3489 audio_track_cblk_t* cblk = this->cblk(); 3490 3491 cblk->user = 0; 3492 cblk->server = 0; 3493 cblk->userBase = 0; 3494 cblk->serverBase = 0; 3495 mStepServerFailed = false; 3496 ALOGV("TrackBase::reset"); 3497} 3498 3499int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3500 return (int)mCblk->sampleRate; 3501} 3502 3503void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3504 audio_track_cblk_t* cblk = this->cblk(); 3505 size_t frameSize = cblk->frameSize; 3506 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3507 int8_t *bufferEnd = bufferStart + frames * frameSize; 3508 3509 // Check validity of returned pointer in case the track control block would have been corrupted. 3510 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 3511 "TrackBase::getBuffer buffer out of range:\n" 3512 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 3513 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 3514 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3515 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 3516 3517 return bufferStart; 3518} 3519 3520status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3521{ 3522 mSyncEvents.add(event); 3523 return NO_ERROR; 3524} 3525 3526// ---------------------------------------------------------------------------- 3527 3528// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3529AudioFlinger::PlaybackThread::Track::Track( 3530 PlaybackThread *thread, 3531 const sp<Client>& client, 3532 audio_stream_type_t streamType, 3533 uint32_t sampleRate, 3534 audio_format_t format, 3535 uint32_t channelMask, 3536 int frameCount, 3537 const sp<IMemory>& sharedBuffer, 3538 int sessionId, 3539 IAudioFlinger::track_flags_t flags) 3540 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3541 mMute(false), 3542 // mFillingUpStatus ? 3543 // mRetryCount initialized later when needed 3544 mSharedBuffer(sharedBuffer), 3545 mStreamType(streamType), 3546 mName(-1), // see note below 3547 mMainBuffer(thread->mixBuffer()), 3548 mAuxBuffer(NULL), 3549 mAuxEffectId(0), mHasVolumeController(false), 3550 mPresentationCompleteFrames(0), 3551 mFlags(flags) 3552{ 3553 if (mCblk != NULL) { 3554 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3555 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3556 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3557 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3558 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 3559 if (mName < 0) { 3560 ALOGE("no more track names available"); 3561 } 3562 } 3563 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3564} 3565 3566AudioFlinger::PlaybackThread::Track::~Track() 3567{ 3568 ALOGV("PlaybackThread::Track destructor"); 3569 sp<ThreadBase> thread = mThread.promote(); 3570 if (thread != 0) { 3571 Mutex::Autolock _l(thread->mLock); 3572 mState = TERMINATED; 3573 } 3574} 3575 3576void AudioFlinger::PlaybackThread::Track::destroy() 3577{ 3578 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3579 // by removing it from mTracks vector, so there is a risk that this Tracks's 3580 // destructor is called. As the destructor needs to lock mLock, 3581 // we must acquire a strong reference on this Track before locking mLock 3582 // here so that the destructor is called only when exiting this function. 3583 // On the other hand, as long as Track::destroy() is only called by 3584 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3585 // this Track with its member mTrack. 3586 sp<Track> keep(this); 3587 { // scope for mLock 3588 sp<ThreadBase> thread = mThread.promote(); 3589 if (thread != 0) { 3590 if (!isOutputTrack()) { 3591 if (mState == ACTIVE || mState == RESUMING) { 3592 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3593 3594#ifdef ADD_BATTERY_DATA 3595 // to track the speaker usage 3596 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3597#endif 3598 } 3599 AudioSystem::releaseOutput(thread->id()); 3600 } 3601 Mutex::Autolock _l(thread->mLock); 3602 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3603 playbackThread->destroyTrack_l(this); 3604 } 3605 } 3606} 3607 3608void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3609{ 3610 uint32_t vlr = mCblk->getVolumeLR(); 3611 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", 3612 mName - AudioMixer::TRACK0, 3613 (mClient == 0) ? getpid_cached : mClient->pid(), 3614 mStreamType, 3615 mFormat, 3616 mChannelMask, 3617 mSessionId, 3618 mFrameCount, 3619 mState, 3620 mMute, 3621 mFillingUpStatus, 3622 mCblk->sampleRate, 3623 vlr & 0xFFFF, 3624 vlr >> 16, 3625 mCblk->server, 3626 mCblk->user, 3627 (int)mMainBuffer, 3628 (int)mAuxBuffer); 3629} 3630 3631// AudioBufferProvider interface 3632status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3633 AudioBufferProvider::Buffer* buffer, int64_t pts) 3634{ 3635 audio_track_cblk_t* cblk = this->cblk(); 3636 uint32_t framesReady; 3637 uint32_t framesReq = buffer->frameCount; 3638 3639 // Check if last stepServer failed, try to step now 3640 if (mStepServerFailed) { 3641 if (!step()) goto getNextBuffer_exit; 3642 ALOGV("stepServer recovered"); 3643 mStepServerFailed = false; 3644 } 3645 3646 framesReady = cblk->framesReady(); 3647 3648 if (CC_LIKELY(framesReady)) { 3649 uint32_t s = cblk->server; 3650 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3651 3652 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3653 if (framesReq > framesReady) { 3654 framesReq = framesReady; 3655 } 3656 if (framesReq > bufferEnd - s) { 3657 framesReq = bufferEnd - s; 3658 } 3659 3660 buffer->raw = getBuffer(s, framesReq); 3661 if (buffer->raw == NULL) goto getNextBuffer_exit; 3662 3663 buffer->frameCount = framesReq; 3664 return NO_ERROR; 3665 } 3666 3667getNextBuffer_exit: 3668 buffer->raw = NULL; 3669 buffer->frameCount = 0; 3670 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3671 return NOT_ENOUGH_DATA; 3672} 3673 3674uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3675 return mCblk->framesReady(); 3676} 3677 3678bool AudioFlinger::PlaybackThread::Track::isReady() const { 3679 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3680 3681 if (framesReady() >= mCblk->frameCount || 3682 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3683 mFillingUpStatus = FS_FILLED; 3684 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3685 return true; 3686 } 3687 return false; 3688} 3689 3690status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid, 3691 AudioSystem::sync_event_t event, 3692 int triggerSession) 3693{ 3694 status_t status = NO_ERROR; 3695 ALOGV("start(%d), calling pid %d session %d tid %d", 3696 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3697 // check for use case 2 with missing callback 3698 if (isFastTrack() && (mSharedBuffer == 0) && (tid == 0)) { 3699 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied"); 3700 mFlags &= ~IAudioFlinger::TRACK_FAST; 3701 // FIXME the track must be invalidated and moved to another thread or 3702 // attached directly to the normal mixer now 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 frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4004 / mCblk->frameSize; 4005 int64_t bufEnd; 4006 4007 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4008 &bufEnd)) { 4009 ALOGE("Failed to convert frame count of %lld to media time duration" 4010 " (scale factor %d/%u) in %s", frameCount, 4011 mMediaTimeToSampleTransform.a_to_b_numer, 4012 mMediaTimeToSampleTransform.a_to_b_denom, 4013 __PRETTY_FUNCTION__); 4014 break; 4015 } 4016 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4017 4018 if (bufEnd > mediaTimeNow) 4019 break; 4020 4021 // Is the buffer we want to use in the middle of a mix operation right 4022 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4023 // from the mixer which should be coming back shortly. 4024 if (!trimEnd && mQueueHeadInFlight) { 4025 mTrimQueueHeadOnRelease = true; 4026 } 4027 } 4028 4029 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4030 if (trimStart < trimEnd) { 4031 // Update the bookkeeping for framesReady() 4032 for (size_t i = trimStart; i < trimEnd; ++i) { 4033 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4034 } 4035 4036 // Now actually remove the buffers from the queue. 4037 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4038 } 4039} 4040 4041void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4042 const char* logTag) { 4043 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4044 "%s called (reason \"%s\"), but timed buffer queue has no" 4045 " elements to trim.", __FUNCTION__, logTag); 4046 4047 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4048 mTimedBufferQueue.removeAt(0); 4049} 4050 4051void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4052 const TimedBuffer& buf, 4053 const char* logTag) { 4054 uint32_t bufBytes = buf.buffer()->size(); 4055 uint32_t consumedAlready = buf.position(); 4056 4057 ALOG_ASSERT(consumedAlready <= bufBytes, 4058 "Bad bookkeeping while updating frames pending. Timed buffer is" 4059 " only %u bytes long, but claims to have consumed %u" 4060 " bytes. (update reason: \"%s\")", 4061 bufBytes, consumedAlready, logTag); 4062 4063 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4064 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4065 "Bad bookkeeping while updating frames pending. Should have at" 4066 " least %u queued frames, but we think we have only %u. (update" 4067 " reason: \"%s\")", 4068 bufFrames, mFramesPendingInQueue, logTag); 4069 4070 mFramesPendingInQueue -= bufFrames; 4071} 4072 4073status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4074 const sp<IMemory>& buffer, int64_t pts) { 4075 4076 { 4077 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4078 if (!mMediaTimeTransformValid) 4079 return INVALID_OPERATION; 4080 } 4081 4082 Mutex::Autolock _l(mTimedBufferQueueLock); 4083 4084 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4085 mFramesPendingInQueue += bufFrames; 4086 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4087 4088 return NO_ERROR; 4089} 4090 4091status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4092 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4093 4094 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4095 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4096 target); 4097 4098 if (!(target == TimedAudioTrack::LOCAL_TIME || 4099 target == TimedAudioTrack::COMMON_TIME)) { 4100 return BAD_VALUE; 4101 } 4102 4103 Mutex::Autolock lock(mMediaTimeTransformLock); 4104 mMediaTimeTransform = xform; 4105 mMediaTimeTransformTarget = target; 4106 mMediaTimeTransformValid = true; 4107 4108 return NO_ERROR; 4109} 4110 4111#define min(a, b) ((a) < (b) ? (a) : (b)) 4112 4113// implementation of getNextBuffer for tracks whose buffers have timestamps 4114status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4115 AudioBufferProvider::Buffer* buffer, int64_t pts) 4116{ 4117 if (pts == AudioBufferProvider::kInvalidPTS) { 4118 buffer->raw = 0; 4119 buffer->frameCount = 0; 4120 return INVALID_OPERATION; 4121 } 4122 4123 Mutex::Autolock _l(mTimedBufferQueueLock); 4124 4125 ALOG_ASSERT(!mQueueHeadInFlight, 4126 "getNextBuffer called without releaseBuffer!"); 4127 4128 while (true) { 4129 4130 // if we have no timed buffers, then fail 4131 if (mTimedBufferQueue.isEmpty()) { 4132 buffer->raw = 0; 4133 buffer->frameCount = 0; 4134 return NOT_ENOUGH_DATA; 4135 } 4136 4137 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4138 4139 // calculate the PTS of the head of the timed buffer queue expressed in 4140 // local time 4141 int64_t headLocalPTS; 4142 { 4143 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4144 4145 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4146 4147 if (mMediaTimeTransform.a_to_b_denom == 0) { 4148 // the transform represents a pause, so yield silence 4149 timedYieldSilence_l(buffer->frameCount, buffer); 4150 return NO_ERROR; 4151 } 4152 4153 int64_t transformedPTS; 4154 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4155 &transformedPTS)) { 4156 // the transform failed. this shouldn't happen, but if it does 4157 // then just drop this buffer 4158 ALOGW("timedGetNextBuffer transform failed"); 4159 buffer->raw = 0; 4160 buffer->frameCount = 0; 4161 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4162 return NO_ERROR; 4163 } 4164 4165 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4166 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4167 &headLocalPTS)) { 4168 buffer->raw = 0; 4169 buffer->frameCount = 0; 4170 return INVALID_OPERATION; 4171 } 4172 } else { 4173 headLocalPTS = transformedPTS; 4174 } 4175 } 4176 4177 // adjust the head buffer's PTS to reflect the portion of the head buffer 4178 // that has already been consumed 4179 int64_t effectivePTS = headLocalPTS + 4180 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4181 4182 // Calculate the delta in samples between the head of the input buffer 4183 // queue and the start of the next output buffer that will be written. 4184 // If the transformation fails because of over or underflow, it means 4185 // that the sample's position in the output stream is so far out of 4186 // whack that it should just be dropped. 4187 int64_t sampleDelta; 4188 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4189 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4190 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4191 " mix"); 4192 continue; 4193 } 4194 if (!mLocalTimeToSampleTransform.doForwardTransform( 4195 (effectivePTS - pts) << 32, &sampleDelta)) { 4196 ALOGV("*** too late during sample rate transform: dropped buffer"); 4197 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 4198 continue; 4199 } 4200 4201 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 4202 " sampleDelta=[%d.%08x]", 4203 head.pts(), head.position(), pts, 4204 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 4205 + (sampleDelta >> 32)), 4206 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4207 4208 // if the delta between the ideal placement for the next input sample and 4209 // the current output position is within this threshold, then we will 4210 // concatenate the next input samples to the previous output 4211 const int64_t kSampleContinuityThreshold = 4212 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4213 4214 // if this is the first buffer of audio that we're emitting from this track 4215 // then it should be almost exactly on time. 4216 const int64_t kSampleStartupThreshold = 1LL << 32; 4217 4218 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4219 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4220 // the next input is close enough to being on time, so concatenate it 4221 // with the last output 4222 timedYieldSamples_l(buffer); 4223 4224 ALOGVV("*** on time: head.pos=%d frameCount=%u", 4225 head.position(), buffer->frameCount); 4226 return NO_ERROR; 4227 } else if (sampleDelta > 0) { 4228 // the gap between the current output position and the proper start of 4229 // the next input sample is too big, so fill it with silence 4230 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4231 4232 timedYieldSilence_l(framesUntilNextInput, buffer); 4233 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4234 return NO_ERROR; 4235 } else { 4236 // the next input sample is late 4237 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4238 size_t onTimeSamplePosition = 4239 head.position() + lateFrames * mCblk->frameSize; 4240 4241 if (onTimeSamplePosition > head.buffer()->size()) { 4242 // all the remaining samples in the head are too late, so 4243 // drop it and move on 4244 ALOGV("*** too late: dropped buffer"); 4245 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 4246 continue; 4247 } else { 4248 // skip over the late samples 4249 head.setPosition(onTimeSamplePosition); 4250 4251 // yield the available samples 4252 timedYieldSamples_l(buffer); 4253 4254 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4255 return NO_ERROR; 4256 } 4257 } 4258 } 4259} 4260 4261// Yield samples from the timed buffer queue head up to the given output 4262// buffer's capacity. 4263// 4264// Caller must hold mTimedBufferQueueLock 4265void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 4266 AudioBufferProvider::Buffer* buffer) { 4267 4268 const TimedBuffer& head = mTimedBufferQueue[0]; 4269 4270 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4271 head.position()); 4272 4273 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4274 mCblk->frameSize); 4275 size_t framesRequested = buffer->frameCount; 4276 buffer->frameCount = min(framesLeftInHead, framesRequested); 4277 4278 mQueueHeadInFlight = true; 4279 mTimedAudioOutputOnTime = true; 4280} 4281 4282// Yield samples of silence up to the given output buffer's capacity 4283// 4284// Caller must hold mTimedBufferQueueLock 4285void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 4286 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4287 4288 // lazily allocate a buffer filled with silence 4289 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4290 delete [] mTimedSilenceBuffer; 4291 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4292 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4293 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4294 } 4295 4296 buffer->raw = mTimedSilenceBuffer; 4297 size_t framesRequested = buffer->frameCount; 4298 buffer->frameCount = min(numFrames, framesRequested); 4299 4300 mTimedAudioOutputOnTime = false; 4301} 4302 4303// AudioBufferProvider interface 4304void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4305 AudioBufferProvider::Buffer* buffer) { 4306 4307 Mutex::Autolock _l(mTimedBufferQueueLock); 4308 4309 // If the buffer which was just released is part of the buffer at the head 4310 // of the queue, be sure to update the amt of the buffer which has been 4311 // consumed. If the buffer being returned is not part of the head of the 4312 // queue, its either because the buffer is part of the silence buffer, or 4313 // because the head of the timed queue was trimmed after the mixer called 4314 // getNextBuffer but before the mixer called releaseBuffer. 4315 if (buffer->raw == mTimedSilenceBuffer) { 4316 ALOG_ASSERT(!mQueueHeadInFlight, 4317 "Queue head in flight during release of silence buffer!"); 4318 goto done; 4319 } 4320 4321 ALOG_ASSERT(mQueueHeadInFlight, 4322 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 4323 " head in flight."); 4324 4325 if (mTimedBufferQueue.size()) { 4326 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4327 4328 void* start = head.buffer()->pointer(); 4329 void* end = reinterpret_cast<void*>( 4330 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 4331 + head.buffer()->size()); 4332 4333 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 4334 "released buffer not within the head of the timed buffer" 4335 " queue; qHead = [%p, %p], released buffer = %p", 4336 start, end, buffer->raw); 4337 4338 head.setPosition(head.position() + 4339 (buffer->frameCount * mCblk->frameSize)); 4340 mQueueHeadInFlight = false; 4341 4342 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 4343 "Bad bookkeeping during releaseBuffer! Should have at" 4344 " least %u queued frames, but we think we have only %u", 4345 buffer->frameCount, mFramesPendingInQueue); 4346 4347 mFramesPendingInQueue -= buffer->frameCount; 4348 4349 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 4350 || mTrimQueueHeadOnRelease) { 4351 trimTimedBufferQueueHead_l("releaseBuffer"); 4352 mTrimQueueHeadOnRelease = false; 4353 } 4354 } else { 4355 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 4356 " buffers in the timed buffer queue"); 4357 } 4358 4359done: 4360 buffer->raw = 0; 4361 buffer->frameCount = 0; 4362} 4363 4364uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4365 Mutex::Autolock _l(mTimedBufferQueueLock); 4366 return mFramesPendingInQueue; 4367} 4368 4369AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4370 : mPTS(0), mPosition(0) {} 4371 4372AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4373 const sp<IMemory>& buffer, int64_t pts) 4374 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4375 4376// ---------------------------------------------------------------------------- 4377 4378// RecordTrack constructor must be called with AudioFlinger::mLock held 4379AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4380 RecordThread *thread, 4381 const sp<Client>& client, 4382 uint32_t sampleRate, 4383 audio_format_t format, 4384 uint32_t channelMask, 4385 int frameCount, 4386 int sessionId) 4387 : TrackBase(thread, client, sampleRate, format, 4388 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4389 mOverflow(false) 4390{ 4391 if (mCblk != NULL) { 4392 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4393 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4394 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4395 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4396 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4397 } else { 4398 mCblk->frameSize = sizeof(int8_t); 4399 } 4400 } 4401} 4402 4403AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4404{ 4405 sp<ThreadBase> thread = mThread.promote(); 4406 if (thread != 0) { 4407 AudioSystem::releaseInput(thread->id()); 4408 } 4409} 4410 4411// AudioBufferProvider interface 4412status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4413{ 4414 audio_track_cblk_t* cblk = this->cblk(); 4415 uint32_t framesAvail; 4416 uint32_t framesReq = buffer->frameCount; 4417 4418 // Check if last stepServer failed, try to step now 4419 if (mStepServerFailed) { 4420 if (!step()) goto getNextBuffer_exit; 4421 ALOGV("stepServer recovered"); 4422 mStepServerFailed = false; 4423 } 4424 4425 framesAvail = cblk->framesAvailable_l(); 4426 4427 if (CC_LIKELY(framesAvail)) { 4428 uint32_t s = cblk->server; 4429 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4430 4431 if (framesReq > framesAvail) { 4432 framesReq = framesAvail; 4433 } 4434 if (framesReq > bufferEnd - s) { 4435 framesReq = bufferEnd - s; 4436 } 4437 4438 buffer->raw = getBuffer(s, framesReq); 4439 if (buffer->raw == NULL) goto getNextBuffer_exit; 4440 4441 buffer->frameCount = framesReq; 4442 return NO_ERROR; 4443 } 4444 4445getNextBuffer_exit: 4446 buffer->raw = NULL; 4447 buffer->frameCount = 0; 4448 return NOT_ENOUGH_DATA; 4449} 4450 4451status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid, 4452 AudioSystem::sync_event_t event, 4453 int triggerSession) 4454{ 4455 sp<ThreadBase> thread = mThread.promote(); 4456 if (thread != 0) { 4457 RecordThread *recordThread = (RecordThread *)thread.get(); 4458 return recordThread->start(this, tid, event, triggerSession); 4459 } else { 4460 return BAD_VALUE; 4461 } 4462} 4463 4464void AudioFlinger::RecordThread::RecordTrack::stop() 4465{ 4466 sp<ThreadBase> thread = mThread.promote(); 4467 if (thread != 0) { 4468 RecordThread *recordThread = (RecordThread *)thread.get(); 4469 recordThread->stop(this); 4470 TrackBase::reset(); 4471 // Force overrun condition to avoid false overrun callback until first data is 4472 // read from buffer 4473 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4474 } 4475} 4476 4477void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4478{ 4479 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4480 (mClient == 0) ? getpid_cached : mClient->pid(), 4481 mFormat, 4482 mChannelMask, 4483 mSessionId, 4484 mFrameCount, 4485 mState, 4486 mCblk->sampleRate, 4487 mCblk->server, 4488 mCblk->user); 4489} 4490 4491 4492// ---------------------------------------------------------------------------- 4493 4494AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4495 PlaybackThread *playbackThread, 4496 DuplicatingThread *sourceThread, 4497 uint32_t sampleRate, 4498 audio_format_t format, 4499 uint32_t channelMask, 4500 int frameCount) 4501 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 4502 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 4503 mActive(false), mSourceThread(sourceThread) 4504{ 4505 4506 if (mCblk != NULL) { 4507 mCblk->flags |= CBLK_DIRECTION_OUT; 4508 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4509 mOutBuffer.frameCount = 0; 4510 playbackThread->mTracks.add(this); 4511 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4512 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4513 mCblk, mBuffer, mCblk->buffers, 4514 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4515 } else { 4516 ALOGW("Error creating output track on thread %p", playbackThread); 4517 } 4518} 4519 4520AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4521{ 4522 clearBufferQueue(); 4523} 4524 4525status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid, 4526 AudioSystem::sync_event_t event, 4527 int triggerSession) 4528{ 4529 status_t status = Track::start(tid, event, triggerSession); 4530 if (status != NO_ERROR) { 4531 return status; 4532 } 4533 4534 mActive = true; 4535 mRetryCount = 127; 4536 return status; 4537} 4538 4539void AudioFlinger::PlaybackThread::OutputTrack::stop() 4540{ 4541 Track::stop(); 4542 clearBufferQueue(); 4543 mOutBuffer.frameCount = 0; 4544 mActive = false; 4545} 4546 4547bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4548{ 4549 Buffer *pInBuffer; 4550 Buffer inBuffer; 4551 uint32_t channelCount = mChannelCount; 4552 bool outputBufferFull = false; 4553 inBuffer.frameCount = frames; 4554 inBuffer.i16 = data; 4555 4556 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4557 4558 if (!mActive && frames != 0) { 4559 start(0); 4560 sp<ThreadBase> thread = mThread.promote(); 4561 if (thread != 0) { 4562 MixerThread *mixerThread = (MixerThread *)thread.get(); 4563 if (mCblk->frameCount > frames){ 4564 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4565 uint32_t startFrames = (mCblk->frameCount - frames); 4566 pInBuffer = new Buffer; 4567 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4568 pInBuffer->frameCount = startFrames; 4569 pInBuffer->i16 = pInBuffer->mBuffer; 4570 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4571 mBufferQueue.add(pInBuffer); 4572 } else { 4573 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4574 } 4575 } 4576 } 4577 } 4578 4579 while (waitTimeLeftMs) { 4580 // First write pending buffers, then new data 4581 if (mBufferQueue.size()) { 4582 pInBuffer = mBufferQueue.itemAt(0); 4583 } else { 4584 pInBuffer = &inBuffer; 4585 } 4586 4587 if (pInBuffer->frameCount == 0) { 4588 break; 4589 } 4590 4591 if (mOutBuffer.frameCount == 0) { 4592 mOutBuffer.frameCount = pInBuffer->frameCount; 4593 nsecs_t startTime = systemTime(); 4594 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4595 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4596 outputBufferFull = true; 4597 break; 4598 } 4599 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4600 if (waitTimeLeftMs >= waitTimeMs) { 4601 waitTimeLeftMs -= waitTimeMs; 4602 } else { 4603 waitTimeLeftMs = 0; 4604 } 4605 } 4606 4607 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4608 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4609 mCblk->stepUser(outFrames); 4610 pInBuffer->frameCount -= outFrames; 4611 pInBuffer->i16 += outFrames * channelCount; 4612 mOutBuffer.frameCount -= outFrames; 4613 mOutBuffer.i16 += outFrames * channelCount; 4614 4615 if (pInBuffer->frameCount == 0) { 4616 if (mBufferQueue.size()) { 4617 mBufferQueue.removeAt(0); 4618 delete [] pInBuffer->mBuffer; 4619 delete pInBuffer; 4620 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4621 } else { 4622 break; 4623 } 4624 } 4625 } 4626 4627 // If we could not write all frames, allocate a buffer and queue it for next time. 4628 if (inBuffer.frameCount) { 4629 sp<ThreadBase> thread = mThread.promote(); 4630 if (thread != 0 && !thread->standby()) { 4631 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4632 pInBuffer = new Buffer; 4633 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4634 pInBuffer->frameCount = inBuffer.frameCount; 4635 pInBuffer->i16 = pInBuffer->mBuffer; 4636 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4637 mBufferQueue.add(pInBuffer); 4638 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4639 } else { 4640 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4641 } 4642 } 4643 } 4644 4645 // Calling write() with a 0 length buffer, means that no more data will be written: 4646 // If no more buffers are pending, fill output track buffer to make sure it is started 4647 // by output mixer. 4648 if (frames == 0 && mBufferQueue.size() == 0) { 4649 if (mCblk->user < mCblk->frameCount) { 4650 frames = mCblk->frameCount - mCblk->user; 4651 pInBuffer = new Buffer; 4652 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4653 pInBuffer->frameCount = frames; 4654 pInBuffer->i16 = pInBuffer->mBuffer; 4655 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4656 mBufferQueue.add(pInBuffer); 4657 } else if (mActive) { 4658 stop(); 4659 } 4660 } 4661 4662 return outputBufferFull; 4663} 4664 4665status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4666{ 4667 int active; 4668 status_t result; 4669 audio_track_cblk_t* cblk = mCblk; 4670 uint32_t framesReq = buffer->frameCount; 4671 4672// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4673 buffer->frameCount = 0; 4674 4675 uint32_t framesAvail = cblk->framesAvailable(); 4676 4677 4678 if (framesAvail == 0) { 4679 Mutex::Autolock _l(cblk->lock); 4680 goto start_loop_here; 4681 while (framesAvail == 0) { 4682 active = mActive; 4683 if (CC_UNLIKELY(!active)) { 4684 ALOGV("Not active and NO_MORE_BUFFERS"); 4685 return NO_MORE_BUFFERS; 4686 } 4687 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4688 if (result != NO_ERROR) { 4689 return NO_MORE_BUFFERS; 4690 } 4691 // read the server count again 4692 start_loop_here: 4693 framesAvail = cblk->framesAvailable_l(); 4694 } 4695 } 4696 4697// if (framesAvail < framesReq) { 4698// return NO_MORE_BUFFERS; 4699// } 4700 4701 if (framesReq > framesAvail) { 4702 framesReq = framesAvail; 4703 } 4704 4705 uint32_t u = cblk->user; 4706 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4707 4708 if (framesReq > bufferEnd - u) { 4709 framesReq = bufferEnd - u; 4710 } 4711 4712 buffer->frameCount = framesReq; 4713 buffer->raw = (void *)cblk->buffer(u); 4714 return NO_ERROR; 4715} 4716 4717 4718void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4719{ 4720 size_t size = mBufferQueue.size(); 4721 4722 for (size_t i = 0; i < size; i++) { 4723 Buffer *pBuffer = mBufferQueue.itemAt(i); 4724 delete [] pBuffer->mBuffer; 4725 delete pBuffer; 4726 } 4727 mBufferQueue.clear(); 4728} 4729 4730// ---------------------------------------------------------------------------- 4731 4732AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4733 : RefBase(), 4734 mAudioFlinger(audioFlinger), 4735 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4736 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4737 mPid(pid), 4738 mTimedTrackCount(0) 4739{ 4740 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4741} 4742 4743// Client destructor must be called with AudioFlinger::mLock held 4744AudioFlinger::Client::~Client() 4745{ 4746 mAudioFlinger->removeClient_l(mPid); 4747} 4748 4749sp<MemoryDealer> AudioFlinger::Client::heap() const 4750{ 4751 return mMemoryDealer; 4752} 4753 4754// Reserve one of the limited slots for a timed audio track associated 4755// with this client 4756bool AudioFlinger::Client::reserveTimedTrack() 4757{ 4758 const int kMaxTimedTracksPerClient = 4; 4759 4760 Mutex::Autolock _l(mTimedTrackLock); 4761 4762 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4763 ALOGW("can not create timed track - pid %d has exceeded the limit", 4764 mPid); 4765 return false; 4766 } 4767 4768 mTimedTrackCount++; 4769 return true; 4770} 4771 4772// Release a slot for a timed audio track 4773void AudioFlinger::Client::releaseTimedTrack() 4774{ 4775 Mutex::Autolock _l(mTimedTrackLock); 4776 mTimedTrackCount--; 4777} 4778 4779// ---------------------------------------------------------------------------- 4780 4781AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4782 const sp<IAudioFlingerClient>& client, 4783 pid_t pid) 4784 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4785{ 4786} 4787 4788AudioFlinger::NotificationClient::~NotificationClient() 4789{ 4790} 4791 4792void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4793{ 4794 sp<NotificationClient> keep(this); 4795 mAudioFlinger->removeNotificationClient(mPid); 4796} 4797 4798// ---------------------------------------------------------------------------- 4799 4800AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4801 : BnAudioTrack(), 4802 mTrack(track) 4803{ 4804} 4805 4806AudioFlinger::TrackHandle::~TrackHandle() { 4807 // just stop the track on deletion, associated resources 4808 // will be freed from the main thread once all pending buffers have 4809 // been played. Unless it's not in the active track list, in which 4810 // case we free everything now... 4811 mTrack->destroy(); 4812} 4813 4814sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4815 return mTrack->getCblk(); 4816} 4817 4818status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4819 return mTrack->start(tid); 4820} 4821 4822void AudioFlinger::TrackHandle::stop() { 4823 mTrack->stop(); 4824} 4825 4826void AudioFlinger::TrackHandle::flush() { 4827 mTrack->flush(); 4828} 4829 4830void AudioFlinger::TrackHandle::mute(bool e) { 4831 mTrack->mute(e); 4832} 4833 4834void AudioFlinger::TrackHandle::pause() { 4835 mTrack->pause(); 4836} 4837 4838status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4839{ 4840 return mTrack->attachAuxEffect(EffectId); 4841} 4842 4843status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4844 sp<IMemory>* buffer) { 4845 if (!mTrack->isTimedTrack()) 4846 return INVALID_OPERATION; 4847 4848 PlaybackThread::TimedTrack* tt = 4849 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4850 return tt->allocateTimedBuffer(size, buffer); 4851} 4852 4853status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4854 int64_t pts) { 4855 if (!mTrack->isTimedTrack()) 4856 return INVALID_OPERATION; 4857 4858 PlaybackThread::TimedTrack* tt = 4859 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4860 return tt->queueTimedBuffer(buffer, pts); 4861} 4862 4863status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4864 const LinearTransform& xform, int target) { 4865 4866 if (!mTrack->isTimedTrack()) 4867 return INVALID_OPERATION; 4868 4869 PlaybackThread::TimedTrack* tt = 4870 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4871 return tt->setMediaTimeTransform( 4872 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4873} 4874 4875status_t AudioFlinger::TrackHandle::onTransact( 4876 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4877{ 4878 return BnAudioTrack::onTransact(code, data, reply, flags); 4879} 4880 4881// ---------------------------------------------------------------------------- 4882 4883sp<IAudioRecord> AudioFlinger::openRecord( 4884 pid_t pid, 4885 audio_io_handle_t input, 4886 uint32_t sampleRate, 4887 audio_format_t format, 4888 uint32_t channelMask, 4889 int frameCount, 4890 IAudioFlinger::track_flags_t flags, 4891 int *sessionId, 4892 status_t *status) 4893{ 4894 sp<RecordThread::RecordTrack> recordTrack; 4895 sp<RecordHandle> recordHandle; 4896 sp<Client> client; 4897 status_t lStatus; 4898 RecordThread *thread; 4899 size_t inFrameCount; 4900 int lSessionId; 4901 4902 // check calling permissions 4903 if (!recordingAllowed()) { 4904 lStatus = PERMISSION_DENIED; 4905 goto Exit; 4906 } 4907 4908 // add client to list 4909 { // scope for mLock 4910 Mutex::Autolock _l(mLock); 4911 thread = checkRecordThread_l(input); 4912 if (thread == NULL) { 4913 lStatus = BAD_VALUE; 4914 goto Exit; 4915 } 4916 4917 client = registerPid_l(pid); 4918 4919 // If no audio session id is provided, create one here 4920 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4921 lSessionId = *sessionId; 4922 } else { 4923 lSessionId = nextUniqueId(); 4924 if (sessionId != NULL) { 4925 *sessionId = lSessionId; 4926 } 4927 } 4928 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4929 recordTrack = thread->createRecordTrack_l(client, 4930 sampleRate, 4931 format, 4932 channelMask, 4933 frameCount, 4934 lSessionId, 4935 &lStatus); 4936 } 4937 if (lStatus != NO_ERROR) { 4938 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4939 // destructor is called by the TrackBase destructor with mLock held 4940 client.clear(); 4941 recordTrack.clear(); 4942 goto Exit; 4943 } 4944 4945 // return to handle to client 4946 recordHandle = new RecordHandle(recordTrack); 4947 lStatus = NO_ERROR; 4948 4949Exit: 4950 if (status) { 4951 *status = lStatus; 4952 } 4953 return recordHandle; 4954} 4955 4956// ---------------------------------------------------------------------------- 4957 4958AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4959 : BnAudioRecord(), 4960 mRecordTrack(recordTrack) 4961{ 4962} 4963 4964AudioFlinger::RecordHandle::~RecordHandle() { 4965 stop(); 4966} 4967 4968sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4969 return mRecordTrack->getCblk(); 4970} 4971 4972status_t AudioFlinger::RecordHandle::start(pid_t tid, int event, int triggerSession) { 4973 ALOGV("RecordHandle::start()"); 4974 return mRecordTrack->start(tid, (AudioSystem::sync_event_t)event, triggerSession); 4975} 4976 4977void AudioFlinger::RecordHandle::stop() { 4978 ALOGV("RecordHandle::stop()"); 4979 mRecordTrack->stop(); 4980} 4981 4982status_t AudioFlinger::RecordHandle::onTransact( 4983 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4984{ 4985 return BnAudioRecord::onTransact(code, data, reply, flags); 4986} 4987 4988// ---------------------------------------------------------------------------- 4989 4990AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4991 AudioStreamIn *input, 4992 uint32_t sampleRate, 4993 uint32_t channels, 4994 audio_io_handle_t id, 4995 uint32_t device) : 4996 ThreadBase(audioFlinger, id, device, RECORD), 4997 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4998 // mRsmpInIndex and mInputBytes set by readInputParameters() 4999 mReqChannelCount(popcount(channels)), 5000 mReqSampleRate(sampleRate) 5001 // mBytesRead is only meaningful while active, and so is cleared in start() 5002 // (but might be better to also clear here for dump?) 5003{ 5004 snprintf(mName, kNameLength, "AudioIn_%X", id); 5005 5006 readInputParameters(); 5007} 5008 5009 5010AudioFlinger::RecordThread::~RecordThread() 5011{ 5012 delete[] mRsmpInBuffer; 5013 delete mResampler; 5014 delete[] mRsmpOutBuffer; 5015} 5016 5017void AudioFlinger::RecordThread::onFirstRef() 5018{ 5019 run(mName, PRIORITY_URGENT_AUDIO); 5020} 5021 5022status_t AudioFlinger::RecordThread::readyToRun() 5023{ 5024 status_t status = initCheck(); 5025 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5026 return status; 5027} 5028 5029bool AudioFlinger::RecordThread::threadLoop() 5030{ 5031 AudioBufferProvider::Buffer buffer; 5032 sp<RecordTrack> activeTrack; 5033 Vector< sp<EffectChain> > effectChains; 5034 5035 nsecs_t lastWarning = 0; 5036 5037 acquireWakeLock(); 5038 5039 // start recording 5040 while (!exitPending()) { 5041 5042 processConfigEvents(); 5043 5044 { // scope for mLock 5045 Mutex::Autolock _l(mLock); 5046 checkForNewParameters_l(); 5047 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5048 if (!mStandby) { 5049 mInput->stream->common.standby(&mInput->stream->common); 5050 mStandby = true; 5051 } 5052 5053 if (exitPending()) break; 5054 5055 releaseWakeLock_l(); 5056 ALOGV("RecordThread: loop stopping"); 5057 // go to sleep 5058 mWaitWorkCV.wait(mLock); 5059 ALOGV("RecordThread: loop starting"); 5060 acquireWakeLock_l(); 5061 continue; 5062 } 5063 if (mActiveTrack != 0) { 5064 if (mActiveTrack->mState == TrackBase::PAUSING) { 5065 if (!mStandby) { 5066 mInput->stream->common.standby(&mInput->stream->common); 5067 mStandby = true; 5068 } 5069 mActiveTrack.clear(); 5070 mStartStopCond.broadcast(); 5071 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5072 if (mReqChannelCount != mActiveTrack->channelCount()) { 5073 mActiveTrack.clear(); 5074 mStartStopCond.broadcast(); 5075 } else if (mBytesRead != 0) { 5076 // record start succeeds only if first read from audio input 5077 // succeeds 5078 if (mBytesRead > 0) { 5079 mActiveTrack->mState = TrackBase::ACTIVE; 5080 } else { 5081 mActiveTrack.clear(); 5082 } 5083 mStartStopCond.broadcast(); 5084 } 5085 mStandby = false; 5086 } 5087 } 5088 lockEffectChains_l(effectChains); 5089 } 5090 5091 if (mActiveTrack != 0) { 5092 if (mActiveTrack->mState != TrackBase::ACTIVE && 5093 mActiveTrack->mState != TrackBase::RESUMING) { 5094 unlockEffectChains(effectChains); 5095 usleep(kRecordThreadSleepUs); 5096 continue; 5097 } 5098 for (size_t i = 0; i < effectChains.size(); i ++) { 5099 effectChains[i]->process_l(); 5100 } 5101 5102 buffer.frameCount = mFrameCount; 5103 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5104 size_t framesOut = buffer.frameCount; 5105 if (mResampler == NULL) { 5106 // no resampling 5107 while (framesOut) { 5108 size_t framesIn = mFrameCount - mRsmpInIndex; 5109 if (framesIn) { 5110 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5111 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5112 if (framesIn > framesOut) 5113 framesIn = framesOut; 5114 mRsmpInIndex += framesIn; 5115 framesOut -= framesIn; 5116 if ((int)mChannelCount == mReqChannelCount || 5117 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5118 memcpy(dst, src, framesIn * mFrameSize); 5119 } else { 5120 int16_t *src16 = (int16_t *)src; 5121 int16_t *dst16 = (int16_t *)dst; 5122 if (mChannelCount == 1) { 5123 while (framesIn--) { 5124 *dst16++ = *src16; 5125 *dst16++ = *src16++; 5126 } 5127 } else { 5128 while (framesIn--) { 5129 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5130 src16 += 2; 5131 } 5132 } 5133 } 5134 } 5135 if (framesOut && mFrameCount == mRsmpInIndex) { 5136 if (framesOut == mFrameCount && 5137 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5138 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5139 framesOut = 0; 5140 } else { 5141 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5142 mRsmpInIndex = 0; 5143 } 5144 if (mBytesRead < 0) { 5145 ALOGE("Error reading audio input"); 5146 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5147 // Force input into standby so that it tries to 5148 // recover at next read attempt 5149 mInput->stream->common.standby(&mInput->stream->common); 5150 usleep(kRecordThreadSleepUs); 5151 } 5152 mRsmpInIndex = mFrameCount; 5153 framesOut = 0; 5154 buffer.frameCount = 0; 5155 } 5156 } 5157 } 5158 } else { 5159 // resampling 5160 5161 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5162 // alter output frame count as if we were expecting stereo samples 5163 if (mChannelCount == 1 && mReqChannelCount == 1) { 5164 framesOut >>= 1; 5165 } 5166 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5167 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5168 // are 32 bit aligned which should be always true. 5169 if (mChannelCount == 2 && mReqChannelCount == 1) { 5170 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5171 // the resampler always outputs stereo samples: do post stereo to mono conversion 5172 int16_t *src = (int16_t *)mRsmpOutBuffer; 5173 int16_t *dst = buffer.i16; 5174 while (framesOut--) { 5175 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5176 src += 2; 5177 } 5178 } else { 5179 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5180 } 5181 5182 } 5183 if (mFramestoDrop == 0) { 5184 mActiveTrack->releaseBuffer(&buffer); 5185 } else { 5186 if (mFramestoDrop > 0) { 5187 mFramestoDrop -= buffer.frameCount; 5188 if (mFramestoDrop < 0) { 5189 mFramestoDrop = 0; 5190 } 5191 } 5192 } 5193 mActiveTrack->overflow(); 5194 } 5195 // client isn't retrieving buffers fast enough 5196 else { 5197 if (!mActiveTrack->setOverflow()) { 5198 nsecs_t now = systemTime(); 5199 if ((now - lastWarning) > kWarningThrottleNs) { 5200 ALOGW("RecordThread: buffer overflow"); 5201 lastWarning = now; 5202 } 5203 } 5204 // Release the processor for a while before asking for a new buffer. 5205 // This will give the application more chance to read from the buffer and 5206 // clear the overflow. 5207 usleep(kRecordThreadSleepUs); 5208 } 5209 } 5210 // enable changes in effect chain 5211 unlockEffectChains(effectChains); 5212 effectChains.clear(); 5213 } 5214 5215 if (!mStandby) { 5216 mInput->stream->common.standby(&mInput->stream->common); 5217 } 5218 mActiveTrack.clear(); 5219 5220 mStartStopCond.broadcast(); 5221 5222 releaseWakeLock(); 5223 5224 ALOGV("RecordThread %p exiting", this); 5225 return false; 5226} 5227 5228 5229sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5230 const sp<AudioFlinger::Client>& client, 5231 uint32_t sampleRate, 5232 audio_format_t format, 5233 int channelMask, 5234 int frameCount, 5235 int sessionId, 5236 status_t *status) 5237{ 5238 sp<RecordTrack> track; 5239 status_t lStatus; 5240 5241 lStatus = initCheck(); 5242 if (lStatus != NO_ERROR) { 5243 ALOGE("Audio driver not initialized."); 5244 goto Exit; 5245 } 5246 5247 { // scope for mLock 5248 Mutex::Autolock _l(mLock); 5249 5250 track = new RecordTrack(this, client, sampleRate, 5251 format, channelMask, frameCount, sessionId); 5252 5253 if (track->getCblk() == 0) { 5254 lStatus = NO_MEMORY; 5255 goto Exit; 5256 } 5257 5258 mTrack = track.get(); 5259 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5260 bool suspend = audio_is_bluetooth_sco_device( 5261 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5262 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5263 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5264 } 5265 lStatus = NO_ERROR; 5266 5267Exit: 5268 if (status) { 5269 *status = lStatus; 5270 } 5271 return track; 5272} 5273 5274status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5275 pid_t tid, AudioSystem::sync_event_t event, 5276 int triggerSession) 5277{ 5278 ALOGV("RecordThread::start tid=%d, event %d, triggerSession %d", tid, event, triggerSession); 5279 sp<ThreadBase> strongMe = this; 5280 status_t status = NO_ERROR; 5281 5282 if (event == AudioSystem::SYNC_EVENT_NONE) { 5283 mSyncStartEvent.clear(); 5284 mFramestoDrop = 0; 5285 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5286 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5287 triggerSession, 5288 recordTrack->sessionId(), 5289 syncStartEventCallback, 5290 this); 5291 mFramestoDrop = -1; 5292 } 5293 5294 { 5295 AutoMutex lock(mLock); 5296 if (mActiveTrack != 0) { 5297 if (recordTrack != mActiveTrack.get()) { 5298 status = -EBUSY; 5299 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5300 mActiveTrack->mState = TrackBase::ACTIVE; 5301 } 5302 return status; 5303 } 5304 5305 recordTrack->mState = TrackBase::IDLE; 5306 mActiveTrack = recordTrack; 5307 mLock.unlock(); 5308 status_t status = AudioSystem::startInput(mId); 5309 mLock.lock(); 5310 if (status != NO_ERROR) { 5311 mActiveTrack.clear(); 5312 clearSyncStartEvent(); 5313 return status; 5314 } 5315 mRsmpInIndex = mFrameCount; 5316 mBytesRead = 0; 5317 if (mResampler != NULL) { 5318 mResampler->reset(); 5319 } 5320 mActiveTrack->mState = TrackBase::RESUMING; 5321 // signal thread to start 5322 ALOGV("Signal record thread"); 5323 mWaitWorkCV.signal(); 5324 // do not wait for mStartStopCond if exiting 5325 if (exitPending()) { 5326 mActiveTrack.clear(); 5327 status = INVALID_OPERATION; 5328 goto startError; 5329 } 5330 mStartStopCond.wait(mLock); 5331 if (mActiveTrack == 0) { 5332 ALOGV("Record failed to start"); 5333 status = BAD_VALUE; 5334 goto startError; 5335 } 5336 ALOGV("Record started OK"); 5337 return status; 5338 } 5339startError: 5340 AudioSystem::stopInput(mId); 5341 clearSyncStartEvent(); 5342 return status; 5343} 5344 5345void AudioFlinger::RecordThread::clearSyncStartEvent() 5346{ 5347 if (mSyncStartEvent != 0) { 5348 mSyncStartEvent->cancel(); 5349 } 5350 mSyncStartEvent.clear(); 5351} 5352 5353void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5354{ 5355 sp<SyncEvent> strongEvent = event.promote(); 5356 5357 if (strongEvent != 0) { 5358 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5359 me->handleSyncStartEvent(strongEvent); 5360 } 5361} 5362 5363void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5364{ 5365 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5366 mActiveTrack.get(), 5367 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5368 event->listenerSession()); 5369 5370 if (mActiveTrack != 0 && 5371 event == mSyncStartEvent) { 5372 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5373 // from audio HAL 5374 mFramestoDrop = mFrameCount * 2; 5375 mSyncStartEvent.clear(); 5376 } 5377} 5378 5379void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5380 ALOGV("RecordThread::stop"); 5381 sp<ThreadBase> strongMe = this; 5382 { 5383 AutoMutex lock(mLock); 5384 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5385 mActiveTrack->mState = TrackBase::PAUSING; 5386 // do not wait for mStartStopCond if exiting 5387 if (exitPending()) { 5388 return; 5389 } 5390 mStartStopCond.wait(mLock); 5391 // if we have been restarted, recordTrack == mActiveTrack.get() here 5392 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5393 mLock.unlock(); 5394 AudioSystem::stopInput(mId); 5395 mLock.lock(); 5396 ALOGV("Record stopped OK"); 5397 } 5398 } 5399 } 5400} 5401 5402bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5403{ 5404 return false; 5405} 5406 5407status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5408{ 5409 if (!isValidSyncEvent(event)) { 5410 return BAD_VALUE; 5411 } 5412 5413 Mutex::Autolock _l(mLock); 5414 5415 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5416 mTrack->setSyncEvent(event); 5417 return NO_ERROR; 5418 } 5419 return NAME_NOT_FOUND; 5420} 5421 5422status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5423{ 5424 const size_t SIZE = 256; 5425 char buffer[SIZE]; 5426 String8 result; 5427 5428 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5429 result.append(buffer); 5430 5431 if (mActiveTrack != 0) { 5432 result.append("Active Track:\n"); 5433 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5434 mActiveTrack->dump(buffer, SIZE); 5435 result.append(buffer); 5436 5437 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5438 result.append(buffer); 5439 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5440 result.append(buffer); 5441 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5442 result.append(buffer); 5443 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5444 result.append(buffer); 5445 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5446 result.append(buffer); 5447 5448 5449 } else { 5450 result.append("No record client\n"); 5451 } 5452 write(fd, result.string(), result.size()); 5453 5454 dumpBase(fd, args); 5455 dumpEffectChains(fd, args); 5456 5457 return NO_ERROR; 5458} 5459 5460// AudioBufferProvider interface 5461status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5462{ 5463 size_t framesReq = buffer->frameCount; 5464 size_t framesReady = mFrameCount - mRsmpInIndex; 5465 int channelCount; 5466 5467 if (framesReady == 0) { 5468 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5469 if (mBytesRead < 0) { 5470 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5471 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5472 // Force input into standby so that it tries to 5473 // recover at next read attempt 5474 mInput->stream->common.standby(&mInput->stream->common); 5475 usleep(kRecordThreadSleepUs); 5476 } 5477 buffer->raw = NULL; 5478 buffer->frameCount = 0; 5479 return NOT_ENOUGH_DATA; 5480 } 5481 mRsmpInIndex = 0; 5482 framesReady = mFrameCount; 5483 } 5484 5485 if (framesReq > framesReady) { 5486 framesReq = framesReady; 5487 } 5488 5489 if (mChannelCount == 1 && mReqChannelCount == 2) { 5490 channelCount = 1; 5491 } else { 5492 channelCount = 2; 5493 } 5494 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5495 buffer->frameCount = framesReq; 5496 return NO_ERROR; 5497} 5498 5499// AudioBufferProvider interface 5500void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5501{ 5502 mRsmpInIndex += buffer->frameCount; 5503 buffer->frameCount = 0; 5504} 5505 5506bool AudioFlinger::RecordThread::checkForNewParameters_l() 5507{ 5508 bool reconfig = false; 5509 5510 while (!mNewParameters.isEmpty()) { 5511 status_t status = NO_ERROR; 5512 String8 keyValuePair = mNewParameters[0]; 5513 AudioParameter param = AudioParameter(keyValuePair); 5514 int value; 5515 audio_format_t reqFormat = mFormat; 5516 int reqSamplingRate = mReqSampleRate; 5517 int reqChannelCount = mReqChannelCount; 5518 5519 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5520 reqSamplingRate = value; 5521 reconfig = true; 5522 } 5523 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5524 reqFormat = (audio_format_t) value; 5525 reconfig = true; 5526 } 5527 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5528 reqChannelCount = popcount(value); 5529 reconfig = true; 5530 } 5531 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5532 // do not accept frame count changes if tracks are open as the track buffer 5533 // size depends on frame count and correct behavior would not be guaranteed 5534 // if frame count is changed after track creation 5535 if (mActiveTrack != 0) { 5536 status = INVALID_OPERATION; 5537 } else { 5538 reconfig = true; 5539 } 5540 } 5541 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5542 // forward device change to effects that have requested to be 5543 // aware of attached audio device. 5544 for (size_t i = 0; i < mEffectChains.size(); i++) { 5545 mEffectChains[i]->setDevice_l(value); 5546 } 5547 // store input device and output device but do not forward output device to audio HAL. 5548 // Note that status is ignored by the caller for output device 5549 // (see AudioFlinger::setParameters() 5550 if (value & AUDIO_DEVICE_OUT_ALL) { 5551 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5552 status = BAD_VALUE; 5553 } else { 5554 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5555 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5556 if (mTrack != NULL) { 5557 bool suspend = audio_is_bluetooth_sco_device( 5558 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5559 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5560 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5561 } 5562 } 5563 mDevice |= (uint32_t)value; 5564 } 5565 if (status == NO_ERROR) { 5566 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5567 if (status == INVALID_OPERATION) { 5568 mInput->stream->common.standby(&mInput->stream->common); 5569 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5570 keyValuePair.string()); 5571 } 5572 if (reconfig) { 5573 if (status == BAD_VALUE && 5574 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5575 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5576 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5577 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5578 (reqChannelCount <= FCC_2)) { 5579 status = NO_ERROR; 5580 } 5581 if (status == NO_ERROR) { 5582 readInputParameters(); 5583 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5584 } 5585 } 5586 } 5587 5588 mNewParameters.removeAt(0); 5589 5590 mParamStatus = status; 5591 mParamCond.signal(); 5592 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5593 // already timed out waiting for the status and will never signal the condition. 5594 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5595 } 5596 return reconfig; 5597} 5598 5599String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5600{ 5601 char *s; 5602 String8 out_s8 = String8(); 5603 5604 Mutex::Autolock _l(mLock); 5605 if (initCheck() != NO_ERROR) { 5606 return out_s8; 5607 } 5608 5609 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5610 out_s8 = String8(s); 5611 free(s); 5612 return out_s8; 5613} 5614 5615void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5616 AudioSystem::OutputDescriptor desc; 5617 void *param2 = NULL; 5618 5619 switch (event) { 5620 case AudioSystem::INPUT_OPENED: 5621 case AudioSystem::INPUT_CONFIG_CHANGED: 5622 desc.channels = mChannelMask; 5623 desc.samplingRate = mSampleRate; 5624 desc.format = mFormat; 5625 desc.frameCount = mFrameCount; 5626 desc.latency = 0; 5627 param2 = &desc; 5628 break; 5629 5630 case AudioSystem::INPUT_CLOSED: 5631 default: 5632 break; 5633 } 5634 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5635} 5636 5637void AudioFlinger::RecordThread::readInputParameters() 5638{ 5639 delete mRsmpInBuffer; 5640 // mRsmpInBuffer is always assigned a new[] below 5641 delete mRsmpOutBuffer; 5642 mRsmpOutBuffer = NULL; 5643 delete mResampler; 5644 mResampler = NULL; 5645 5646 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5647 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5648 mChannelCount = (uint16_t)popcount(mChannelMask); 5649 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5650 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5651 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5652 mFrameCount = mInputBytes / mFrameSize; 5653 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5654 5655 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5656 { 5657 int channelCount; 5658 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5659 // stereo to mono post process as the resampler always outputs stereo. 5660 if (mChannelCount == 1 && mReqChannelCount == 2) { 5661 channelCount = 1; 5662 } else { 5663 channelCount = 2; 5664 } 5665 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5666 mResampler->setSampleRate(mSampleRate); 5667 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5668 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5669 5670 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5671 if (mChannelCount == 1 && mReqChannelCount == 1) { 5672 mFrameCount >>= 1; 5673 } 5674 5675 } 5676 mRsmpInIndex = mFrameCount; 5677} 5678 5679unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5680{ 5681 Mutex::Autolock _l(mLock); 5682 if (initCheck() != NO_ERROR) { 5683 return 0; 5684 } 5685 5686 return mInput->stream->get_input_frames_lost(mInput->stream); 5687} 5688 5689uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5690{ 5691 Mutex::Autolock _l(mLock); 5692 uint32_t result = 0; 5693 if (getEffectChain_l(sessionId) != 0) { 5694 result = EFFECT_SESSION; 5695 } 5696 5697 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5698 result |= TRACK_SESSION; 5699 } 5700 5701 return result; 5702} 5703 5704AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5705{ 5706 Mutex::Autolock _l(mLock); 5707 return mTrack; 5708} 5709 5710AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5711{ 5712 Mutex::Autolock _l(mLock); 5713 return mInput; 5714} 5715 5716AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5717{ 5718 Mutex::Autolock _l(mLock); 5719 AudioStreamIn *input = mInput; 5720 mInput = NULL; 5721 return input; 5722} 5723 5724// this method must always be called either with ThreadBase mLock held or inside the thread loop 5725audio_stream_t* AudioFlinger::RecordThread::stream() const 5726{ 5727 if (mInput == NULL) { 5728 return NULL; 5729 } 5730 return &mInput->stream->common; 5731} 5732 5733 5734// ---------------------------------------------------------------------------- 5735 5736audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 5737{ 5738 if (!settingsAllowed()) { 5739 return 0; 5740 } 5741 Mutex::Autolock _l(mLock); 5742 return loadHwModule_l(name); 5743} 5744 5745// loadHwModule_l() must be called with AudioFlinger::mLock held 5746audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 5747{ 5748 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5749 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 5750 ALOGW("loadHwModule() module %s already loaded", name); 5751 return mAudioHwDevs.keyAt(i); 5752 } 5753 } 5754 5755 audio_hw_device_t *dev; 5756 5757 int rc = load_audio_interface(name, &dev); 5758 if (rc) { 5759 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 5760 return 0; 5761 } 5762 5763 mHardwareStatus = AUDIO_HW_INIT; 5764 rc = dev->init_check(dev); 5765 mHardwareStatus = AUDIO_HW_IDLE; 5766 if (rc) { 5767 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 5768 return 0; 5769 } 5770 5771 if ((mMasterVolumeSupportLvl != MVS_NONE) && 5772 (NULL != dev->set_master_volume)) { 5773 AutoMutex lock(mHardwareLock); 5774 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5775 dev->set_master_volume(dev, mMasterVolume); 5776 mHardwareStatus = AUDIO_HW_IDLE; 5777 } 5778 5779 audio_module_handle_t handle = nextUniqueId(); 5780 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 5781 5782 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 5783 name, dev->common.module->name, dev->common.module->id, handle); 5784 5785 return handle; 5786 5787} 5788 5789audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 5790 audio_devices_t *pDevices, 5791 uint32_t *pSamplingRate, 5792 audio_format_t *pFormat, 5793 audio_channel_mask_t *pChannelMask, 5794 uint32_t *pLatencyMs, 5795 audio_output_flags_t flags) 5796{ 5797 status_t status; 5798 PlaybackThread *thread = NULL; 5799 struct audio_config config = { 5800 sample_rate: pSamplingRate ? *pSamplingRate : 0, 5801 channel_mask: pChannelMask ? *pChannelMask : 0, 5802 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 5803 }; 5804 audio_stream_out_t *outStream = NULL; 5805 audio_hw_device_t *outHwDev; 5806 5807 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5808 module, 5809 (pDevices != NULL) ? (int)*pDevices : 0, 5810 config.sample_rate, 5811 config.format, 5812 config.channel_mask, 5813 flags); 5814 5815 if (pDevices == NULL || *pDevices == 0) { 5816 return 0; 5817 } 5818 5819 Mutex::Autolock _l(mLock); 5820 5821 outHwDev = findSuitableHwDev_l(module, *pDevices); 5822 if (outHwDev == NULL) 5823 return 0; 5824 5825 audio_io_handle_t id = nextUniqueId(); 5826 5827 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5828 5829 status = outHwDev->open_output_stream(outHwDev, 5830 id, 5831 *pDevices, 5832 (audio_output_flags_t)flags, 5833 &config, 5834 &outStream); 5835 5836 mHardwareStatus = AUDIO_HW_IDLE; 5837 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5838 outStream, 5839 config.sample_rate, 5840 config.format, 5841 config.channel_mask, 5842 status); 5843 5844 if (status == NO_ERROR && outStream != NULL) { 5845 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5846 5847 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 5848 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 5849 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 5850 thread = new DirectOutputThread(this, output, id, *pDevices); 5851 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5852 } else { 5853 thread = new MixerThread(this, output, id, *pDevices); 5854 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5855 } 5856 mPlaybackThreads.add(id, thread); 5857 5858 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 5859 if (pFormat != NULL) *pFormat = config.format; 5860 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 5861 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5862 5863 // notify client processes of the new output creation 5864 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5865 5866 // the first primary output opened designates the primary hw device 5867 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 5868 ALOGI("Using module %d has the primary audio interface", module); 5869 mPrimaryHardwareDev = outHwDev; 5870 5871 AutoMutex lock(mHardwareLock); 5872 mHardwareStatus = AUDIO_HW_SET_MODE; 5873 outHwDev->set_mode(outHwDev, mMode); 5874 5875 // Determine the level of master volume support the primary audio HAL has, 5876 // and set the initial master volume at the same time. 5877 float initialVolume = 1.0; 5878 mMasterVolumeSupportLvl = MVS_NONE; 5879 5880 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 5881 if ((NULL != outHwDev->get_master_volume) && 5882 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 5883 mMasterVolumeSupportLvl = MVS_FULL; 5884 } else { 5885 mMasterVolumeSupportLvl = MVS_SETONLY; 5886 initialVolume = 1.0; 5887 } 5888 5889 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5890 if ((NULL == outHwDev->set_master_volume) || 5891 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 5892 mMasterVolumeSupportLvl = MVS_NONE; 5893 } 5894 // now that we have a primary device, initialize master volume on other devices 5895 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5896 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 5897 5898 if ((dev != mPrimaryHardwareDev) && 5899 (NULL != dev->set_master_volume)) { 5900 dev->set_master_volume(dev, initialVolume); 5901 } 5902 } 5903 mHardwareStatus = AUDIO_HW_IDLE; 5904 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 5905 ? initialVolume 5906 : 1.0; 5907 mMasterVolume = initialVolume; 5908 } 5909 return id; 5910 } 5911 5912 return 0; 5913} 5914 5915audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5916 audio_io_handle_t output2) 5917{ 5918 Mutex::Autolock _l(mLock); 5919 MixerThread *thread1 = checkMixerThread_l(output1); 5920 MixerThread *thread2 = checkMixerThread_l(output2); 5921 5922 if (thread1 == NULL || thread2 == NULL) { 5923 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5924 return 0; 5925 } 5926 5927 audio_io_handle_t id = nextUniqueId(); 5928 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5929 thread->addOutputTrack(thread2); 5930 mPlaybackThreads.add(id, thread); 5931 // notify client processes of the new output creation 5932 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5933 return id; 5934} 5935 5936status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5937{ 5938 // keep strong reference on the playback thread so that 5939 // it is not destroyed while exit() is executed 5940 sp<PlaybackThread> thread; 5941 { 5942 Mutex::Autolock _l(mLock); 5943 thread = checkPlaybackThread_l(output); 5944 if (thread == NULL) { 5945 return BAD_VALUE; 5946 } 5947 5948 ALOGV("closeOutput() %d", output); 5949 5950 if (thread->type() == ThreadBase::MIXER) { 5951 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5952 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5953 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5954 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5955 } 5956 } 5957 } 5958 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5959 mPlaybackThreads.removeItem(output); 5960 } 5961 thread->exit(); 5962 // The thread entity (active unit of execution) is no longer running here, 5963 // but the ThreadBase container still exists. 5964 5965 if (thread->type() != ThreadBase::DUPLICATING) { 5966 AudioStreamOut *out = thread->clearOutput(); 5967 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5968 // from now on thread->mOutput is NULL 5969 out->hwDev->close_output_stream(out->hwDev, out->stream); 5970 delete out; 5971 } 5972 return NO_ERROR; 5973} 5974 5975status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5976{ 5977 Mutex::Autolock _l(mLock); 5978 PlaybackThread *thread = checkPlaybackThread_l(output); 5979 5980 if (thread == NULL) { 5981 return BAD_VALUE; 5982 } 5983 5984 ALOGV("suspendOutput() %d", output); 5985 thread->suspend(); 5986 5987 return NO_ERROR; 5988} 5989 5990status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5991{ 5992 Mutex::Autolock _l(mLock); 5993 PlaybackThread *thread = checkPlaybackThread_l(output); 5994 5995 if (thread == NULL) { 5996 return BAD_VALUE; 5997 } 5998 5999 ALOGV("restoreOutput() %d", output); 6000 6001 thread->restore(); 6002 6003 return NO_ERROR; 6004} 6005 6006audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6007 audio_devices_t *pDevices, 6008 uint32_t *pSamplingRate, 6009 audio_format_t *pFormat, 6010 uint32_t *pChannelMask) 6011{ 6012 status_t status; 6013 RecordThread *thread = NULL; 6014 struct audio_config config = { 6015 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6016 channel_mask: pChannelMask ? *pChannelMask : 0, 6017 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6018 }; 6019 uint32_t reqSamplingRate = config.sample_rate; 6020 audio_format_t reqFormat = config.format; 6021 audio_channel_mask_t reqChannels = config.channel_mask; 6022 audio_stream_in_t *inStream = NULL; 6023 audio_hw_device_t *inHwDev; 6024 6025 if (pDevices == NULL || *pDevices == 0) { 6026 return 0; 6027 } 6028 6029 Mutex::Autolock _l(mLock); 6030 6031 inHwDev = findSuitableHwDev_l(module, *pDevices); 6032 if (inHwDev == NULL) 6033 return 0; 6034 6035 audio_io_handle_t id = nextUniqueId(); 6036 6037 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6038 &inStream); 6039 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6040 inStream, 6041 config.sample_rate, 6042 config.format, 6043 config.channel_mask, 6044 status); 6045 6046 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6047 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6048 // or stereo to mono conversions on 16 bit PCM inputs. 6049 if (status == BAD_VALUE && 6050 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6051 (config.sample_rate <= 2 * reqSamplingRate) && 6052 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6053 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6054 inStream = NULL; 6055 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6056 } 6057 6058 if (status == NO_ERROR && inStream != NULL) { 6059 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6060 6061 // Start record thread 6062 // RecorThread require both input and output device indication to forward to audio 6063 // pre processing modules 6064 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6065 thread = new RecordThread(this, 6066 input, 6067 reqSamplingRate, 6068 reqChannels, 6069 id, 6070 device); 6071 mRecordThreads.add(id, thread); 6072 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6073 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6074 if (pFormat != NULL) *pFormat = config.format; 6075 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6076 6077 input->stream->common.standby(&input->stream->common); 6078 6079 // notify client processes of the new input creation 6080 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6081 return id; 6082 } 6083 6084 return 0; 6085} 6086 6087status_t AudioFlinger::closeInput(audio_io_handle_t input) 6088{ 6089 // keep strong reference on the record thread so that 6090 // it is not destroyed while exit() is executed 6091 sp<RecordThread> thread; 6092 { 6093 Mutex::Autolock _l(mLock); 6094 thread = checkRecordThread_l(input); 6095 if (thread == NULL) { 6096 return BAD_VALUE; 6097 } 6098 6099 ALOGV("closeInput() %d", input); 6100 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6101 mRecordThreads.removeItem(input); 6102 } 6103 thread->exit(); 6104 // The thread entity (active unit of execution) is no longer running here, 6105 // but the ThreadBase container still exists. 6106 6107 AudioStreamIn *in = thread->clearInput(); 6108 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6109 // from now on thread->mInput is NULL 6110 in->hwDev->close_input_stream(in->hwDev, in->stream); 6111 delete in; 6112 6113 return NO_ERROR; 6114} 6115 6116status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6117{ 6118 Mutex::Autolock _l(mLock); 6119 MixerThread *dstThread = checkMixerThread_l(output); 6120 if (dstThread == NULL) { 6121 ALOGW("setStreamOutput() bad output id %d", output); 6122 return BAD_VALUE; 6123 } 6124 6125 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6126 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6127 6128 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6129 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6130 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6131 MixerThread *srcThread = (MixerThread *)thread; 6132 srcThread->invalidateTracks(stream); 6133 } 6134 } 6135 6136 return NO_ERROR; 6137} 6138 6139 6140int AudioFlinger::newAudioSessionId() 6141{ 6142 return nextUniqueId(); 6143} 6144 6145void AudioFlinger::acquireAudioSessionId(int audioSession) 6146{ 6147 Mutex::Autolock _l(mLock); 6148 pid_t caller = IPCThreadState::self()->getCallingPid(); 6149 ALOGV("acquiring %d from %d", audioSession, caller); 6150 size_t num = mAudioSessionRefs.size(); 6151 for (size_t i = 0; i< num; i++) { 6152 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6153 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6154 ref->mCnt++; 6155 ALOGV(" incremented refcount to %d", ref->mCnt); 6156 return; 6157 } 6158 } 6159 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6160 ALOGV(" added new entry for %d", audioSession); 6161} 6162 6163void AudioFlinger::releaseAudioSessionId(int audioSession) 6164{ 6165 Mutex::Autolock _l(mLock); 6166 pid_t caller = IPCThreadState::self()->getCallingPid(); 6167 ALOGV("releasing %d from %d", audioSession, caller); 6168 size_t num = mAudioSessionRefs.size(); 6169 for (size_t i = 0; i< num; i++) { 6170 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6171 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6172 ref->mCnt--; 6173 ALOGV(" decremented refcount to %d", ref->mCnt); 6174 if (ref->mCnt == 0) { 6175 mAudioSessionRefs.removeAt(i); 6176 delete ref; 6177 purgeStaleEffects_l(); 6178 } 6179 return; 6180 } 6181 } 6182 ALOGW("session id %d not found for pid %d", audioSession, caller); 6183} 6184 6185void AudioFlinger::purgeStaleEffects_l() { 6186 6187 ALOGV("purging stale effects"); 6188 6189 Vector< sp<EffectChain> > chains; 6190 6191 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6192 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6193 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6194 sp<EffectChain> ec = t->mEffectChains[j]; 6195 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6196 chains.push(ec); 6197 } 6198 } 6199 } 6200 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6201 sp<RecordThread> t = mRecordThreads.valueAt(i); 6202 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6203 sp<EffectChain> ec = t->mEffectChains[j]; 6204 chains.push(ec); 6205 } 6206 } 6207 6208 for (size_t i = 0; i < chains.size(); i++) { 6209 sp<EffectChain> ec = chains[i]; 6210 int sessionid = ec->sessionId(); 6211 sp<ThreadBase> t = ec->mThread.promote(); 6212 if (t == 0) { 6213 continue; 6214 } 6215 size_t numsessionrefs = mAudioSessionRefs.size(); 6216 bool found = false; 6217 for (size_t k = 0; k < numsessionrefs; k++) { 6218 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6219 if (ref->mSessionid == sessionid) { 6220 ALOGV(" session %d still exists for %d with %d refs", 6221 sessionid, ref->mPid, ref->mCnt); 6222 found = true; 6223 break; 6224 } 6225 } 6226 if (!found) { 6227 // remove all effects from the chain 6228 while (ec->mEffects.size()) { 6229 sp<EffectModule> effect = ec->mEffects[0]; 6230 effect->unPin(); 6231 Mutex::Autolock _l (t->mLock); 6232 t->removeEffect_l(effect); 6233 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6234 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6235 if (handle != 0) { 6236 handle->mEffect.clear(); 6237 if (handle->mHasControl && handle->mEnabled) { 6238 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6239 } 6240 } 6241 } 6242 AudioSystem::unregisterEffect(effect->id()); 6243 } 6244 } 6245 } 6246 return; 6247} 6248 6249// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6250AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6251{ 6252 return mPlaybackThreads.valueFor(output).get(); 6253} 6254 6255// checkMixerThread_l() must be called with AudioFlinger::mLock held 6256AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6257{ 6258 PlaybackThread *thread = checkPlaybackThread_l(output); 6259 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6260} 6261 6262// checkRecordThread_l() must be called with AudioFlinger::mLock held 6263AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6264{ 6265 return mRecordThreads.valueFor(input).get(); 6266} 6267 6268uint32_t AudioFlinger::nextUniqueId() 6269{ 6270 return android_atomic_inc(&mNextUniqueId); 6271} 6272 6273AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6274{ 6275 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6276 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6277 AudioStreamOut *output = thread->getOutput(); 6278 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6279 return thread; 6280 } 6281 } 6282 return NULL; 6283} 6284 6285uint32_t AudioFlinger::primaryOutputDevice_l() const 6286{ 6287 PlaybackThread *thread = primaryPlaybackThread_l(); 6288 6289 if (thread == NULL) { 6290 return 0; 6291 } 6292 6293 return thread->device(); 6294} 6295 6296sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6297 int triggerSession, 6298 int listenerSession, 6299 sync_event_callback_t callBack, 6300 void *cookie) 6301{ 6302 Mutex::Autolock _l(mLock); 6303 6304 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6305 status_t playStatus = NAME_NOT_FOUND; 6306 status_t recStatus = NAME_NOT_FOUND; 6307 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6308 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6309 if (playStatus == NO_ERROR) { 6310 return event; 6311 } 6312 } 6313 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6314 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6315 if (recStatus == NO_ERROR) { 6316 return event; 6317 } 6318 } 6319 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6320 mPendingSyncEvents.add(event); 6321 } else { 6322 ALOGV("createSyncEvent() invalid event %d", event->type()); 6323 event.clear(); 6324 } 6325 return event; 6326} 6327 6328// ---------------------------------------------------------------------------- 6329// Effect management 6330// ---------------------------------------------------------------------------- 6331 6332 6333status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6334{ 6335 Mutex::Autolock _l(mLock); 6336 return EffectQueryNumberEffects(numEffects); 6337} 6338 6339status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6340{ 6341 Mutex::Autolock _l(mLock); 6342 return EffectQueryEffect(index, descriptor); 6343} 6344 6345status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6346 effect_descriptor_t *descriptor) const 6347{ 6348 Mutex::Autolock _l(mLock); 6349 return EffectGetDescriptor(pUuid, descriptor); 6350} 6351 6352 6353sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6354 effect_descriptor_t *pDesc, 6355 const sp<IEffectClient>& effectClient, 6356 int32_t priority, 6357 audio_io_handle_t io, 6358 int sessionId, 6359 status_t *status, 6360 int *id, 6361 int *enabled) 6362{ 6363 status_t lStatus = NO_ERROR; 6364 sp<EffectHandle> handle; 6365 effect_descriptor_t desc; 6366 6367 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6368 pid, effectClient.get(), priority, sessionId, io); 6369 6370 if (pDesc == NULL) { 6371 lStatus = BAD_VALUE; 6372 goto Exit; 6373 } 6374 6375 // check audio settings permission for global effects 6376 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6377 lStatus = PERMISSION_DENIED; 6378 goto Exit; 6379 } 6380 6381 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6382 // that can only be created by audio policy manager (running in same process) 6383 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6384 lStatus = PERMISSION_DENIED; 6385 goto Exit; 6386 } 6387 6388 if (io == 0) { 6389 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6390 // output must be specified by AudioPolicyManager when using session 6391 // AUDIO_SESSION_OUTPUT_STAGE 6392 lStatus = BAD_VALUE; 6393 goto Exit; 6394 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6395 // if the output returned by getOutputForEffect() is removed before we lock the 6396 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6397 // and we will exit safely 6398 io = AudioSystem::getOutputForEffect(&desc); 6399 } 6400 } 6401 6402 { 6403 Mutex::Autolock _l(mLock); 6404 6405 6406 if (!EffectIsNullUuid(&pDesc->uuid)) { 6407 // if uuid is specified, request effect descriptor 6408 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6409 if (lStatus < 0) { 6410 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6411 goto Exit; 6412 } 6413 } else { 6414 // if uuid is not specified, look for an available implementation 6415 // of the required type in effect factory 6416 if (EffectIsNullUuid(&pDesc->type)) { 6417 ALOGW("createEffect() no effect type"); 6418 lStatus = BAD_VALUE; 6419 goto Exit; 6420 } 6421 uint32_t numEffects = 0; 6422 effect_descriptor_t d; 6423 d.flags = 0; // prevent compiler warning 6424 bool found = false; 6425 6426 lStatus = EffectQueryNumberEffects(&numEffects); 6427 if (lStatus < 0) { 6428 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6429 goto Exit; 6430 } 6431 for (uint32_t i = 0; i < numEffects; i++) { 6432 lStatus = EffectQueryEffect(i, &desc); 6433 if (lStatus < 0) { 6434 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6435 continue; 6436 } 6437 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6438 // If matching type found save effect descriptor. If the session is 6439 // 0 and the effect is not auxiliary, continue enumeration in case 6440 // an auxiliary version of this effect type is available 6441 found = true; 6442 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6443 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6444 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6445 break; 6446 } 6447 } 6448 } 6449 if (!found) { 6450 lStatus = BAD_VALUE; 6451 ALOGW("createEffect() effect not found"); 6452 goto Exit; 6453 } 6454 // For same effect type, chose auxiliary version over insert version if 6455 // connect to output mix (Compliance to OpenSL ES) 6456 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6457 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6458 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6459 } 6460 } 6461 6462 // Do not allow auxiliary effects on a session different from 0 (output mix) 6463 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6464 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6465 lStatus = INVALID_OPERATION; 6466 goto Exit; 6467 } 6468 6469 // check recording permission for visualizer 6470 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6471 !recordingAllowed()) { 6472 lStatus = PERMISSION_DENIED; 6473 goto Exit; 6474 } 6475 6476 // return effect descriptor 6477 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6478 6479 // If output is not specified try to find a matching audio session ID in one of the 6480 // output threads. 6481 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6482 // because of code checking output when entering the function. 6483 // Note: io is never 0 when creating an effect on an input 6484 if (io == 0) { 6485 // look for the thread where the specified audio session is present 6486 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6487 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6488 io = mPlaybackThreads.keyAt(i); 6489 break; 6490 } 6491 } 6492 if (io == 0) { 6493 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6494 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6495 io = mRecordThreads.keyAt(i); 6496 break; 6497 } 6498 } 6499 } 6500 // If no output thread contains the requested session ID, default to 6501 // first output. The effect chain will be moved to the correct output 6502 // thread when a track with the same session ID is created 6503 if (io == 0 && mPlaybackThreads.size()) { 6504 io = mPlaybackThreads.keyAt(0); 6505 } 6506 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6507 } 6508 ThreadBase *thread = checkRecordThread_l(io); 6509 if (thread == NULL) { 6510 thread = checkPlaybackThread_l(io); 6511 if (thread == NULL) { 6512 ALOGE("createEffect() unknown output thread"); 6513 lStatus = BAD_VALUE; 6514 goto Exit; 6515 } 6516 } 6517 6518 sp<Client> client = registerPid_l(pid); 6519 6520 // create effect on selected output thread 6521 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6522 &desc, enabled, &lStatus); 6523 if (handle != 0 && id != NULL) { 6524 *id = handle->id(); 6525 } 6526 } 6527 6528Exit: 6529 if (status != NULL) { 6530 *status = lStatus; 6531 } 6532 return handle; 6533} 6534 6535status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6536 audio_io_handle_t dstOutput) 6537{ 6538 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6539 sessionId, srcOutput, dstOutput); 6540 Mutex::Autolock _l(mLock); 6541 if (srcOutput == dstOutput) { 6542 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6543 return NO_ERROR; 6544 } 6545 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6546 if (srcThread == NULL) { 6547 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6548 return BAD_VALUE; 6549 } 6550 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6551 if (dstThread == NULL) { 6552 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6553 return BAD_VALUE; 6554 } 6555 6556 Mutex::Autolock _dl(dstThread->mLock); 6557 Mutex::Autolock _sl(srcThread->mLock); 6558 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6559 6560 return NO_ERROR; 6561} 6562 6563// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6564status_t AudioFlinger::moveEffectChain_l(int sessionId, 6565 AudioFlinger::PlaybackThread *srcThread, 6566 AudioFlinger::PlaybackThread *dstThread, 6567 bool reRegister) 6568{ 6569 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6570 sessionId, srcThread, dstThread); 6571 6572 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6573 if (chain == 0) { 6574 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6575 sessionId, srcThread); 6576 return INVALID_OPERATION; 6577 } 6578 6579 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6580 // so that a new chain is created with correct parameters when first effect is added. This is 6581 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6582 // removed. 6583 srcThread->removeEffectChain_l(chain); 6584 6585 // transfer all effects one by one so that new effect chain is created on new thread with 6586 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6587 audio_io_handle_t dstOutput = dstThread->id(); 6588 sp<EffectChain> dstChain; 6589 uint32_t strategy = 0; // prevent compiler warning 6590 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6591 while (effect != 0) { 6592 srcThread->removeEffect_l(effect); 6593 dstThread->addEffect_l(effect); 6594 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6595 if (effect->state() == EffectModule::ACTIVE || 6596 effect->state() == EffectModule::STOPPING) { 6597 effect->start(); 6598 } 6599 // if the move request is not received from audio policy manager, the effect must be 6600 // re-registered with the new strategy and output 6601 if (dstChain == 0) { 6602 dstChain = effect->chain().promote(); 6603 if (dstChain == 0) { 6604 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6605 srcThread->addEffect_l(effect); 6606 return NO_INIT; 6607 } 6608 strategy = dstChain->strategy(); 6609 } 6610 if (reRegister) { 6611 AudioSystem::unregisterEffect(effect->id()); 6612 AudioSystem::registerEffect(&effect->desc(), 6613 dstOutput, 6614 strategy, 6615 sessionId, 6616 effect->id()); 6617 } 6618 effect = chain->getEffectFromId_l(0); 6619 } 6620 6621 return NO_ERROR; 6622} 6623 6624 6625// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6626sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6627 const sp<AudioFlinger::Client>& client, 6628 const sp<IEffectClient>& effectClient, 6629 int32_t priority, 6630 int sessionId, 6631 effect_descriptor_t *desc, 6632 int *enabled, 6633 status_t *status 6634 ) 6635{ 6636 sp<EffectModule> effect; 6637 sp<EffectHandle> handle; 6638 status_t lStatus; 6639 sp<EffectChain> chain; 6640 bool chainCreated = false; 6641 bool effectCreated = false; 6642 bool effectRegistered = false; 6643 6644 lStatus = initCheck(); 6645 if (lStatus != NO_ERROR) { 6646 ALOGW("createEffect_l() Audio driver not initialized."); 6647 goto Exit; 6648 } 6649 6650 // Do not allow effects with session ID 0 on direct output or duplicating threads 6651 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6652 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6653 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6654 desc->name, sessionId); 6655 lStatus = BAD_VALUE; 6656 goto Exit; 6657 } 6658 // Only Pre processor effects are allowed on input threads and only on input threads 6659 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6660 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6661 desc->name, desc->flags, mType); 6662 lStatus = BAD_VALUE; 6663 goto Exit; 6664 } 6665 6666 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6667 6668 { // scope for mLock 6669 Mutex::Autolock _l(mLock); 6670 6671 // check for existing effect chain with the requested audio session 6672 chain = getEffectChain_l(sessionId); 6673 if (chain == 0) { 6674 // create a new chain for this session 6675 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6676 chain = new EffectChain(this, sessionId); 6677 addEffectChain_l(chain); 6678 chain->setStrategy(getStrategyForSession_l(sessionId)); 6679 chainCreated = true; 6680 } else { 6681 effect = chain->getEffectFromDesc_l(desc); 6682 } 6683 6684 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6685 6686 if (effect == 0) { 6687 int id = mAudioFlinger->nextUniqueId(); 6688 // Check CPU and memory usage 6689 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6690 if (lStatus != NO_ERROR) { 6691 goto Exit; 6692 } 6693 effectRegistered = true; 6694 // create a new effect module if none present in the chain 6695 effect = new EffectModule(this, chain, desc, id, sessionId); 6696 lStatus = effect->status(); 6697 if (lStatus != NO_ERROR) { 6698 goto Exit; 6699 } 6700 lStatus = chain->addEffect_l(effect); 6701 if (lStatus != NO_ERROR) { 6702 goto Exit; 6703 } 6704 effectCreated = true; 6705 6706 effect->setDevice(mDevice); 6707 effect->setMode(mAudioFlinger->getMode()); 6708 } 6709 // create effect handle and connect it to effect module 6710 handle = new EffectHandle(effect, client, effectClient, priority); 6711 lStatus = effect->addHandle(handle); 6712 if (enabled != NULL) { 6713 *enabled = (int)effect->isEnabled(); 6714 } 6715 } 6716 6717Exit: 6718 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6719 Mutex::Autolock _l(mLock); 6720 if (effectCreated) { 6721 chain->removeEffect_l(effect); 6722 } 6723 if (effectRegistered) { 6724 AudioSystem::unregisterEffect(effect->id()); 6725 } 6726 if (chainCreated) { 6727 removeEffectChain_l(chain); 6728 } 6729 handle.clear(); 6730 } 6731 6732 if (status != NULL) { 6733 *status = lStatus; 6734 } 6735 return handle; 6736} 6737 6738sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6739{ 6740 sp<EffectChain> chain = getEffectChain_l(sessionId); 6741 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6742} 6743 6744// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6745// PlaybackThread::mLock held 6746status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6747{ 6748 // check for existing effect chain with the requested audio session 6749 int sessionId = effect->sessionId(); 6750 sp<EffectChain> chain = getEffectChain_l(sessionId); 6751 bool chainCreated = false; 6752 6753 if (chain == 0) { 6754 // create a new chain for this session 6755 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6756 chain = new EffectChain(this, sessionId); 6757 addEffectChain_l(chain); 6758 chain->setStrategy(getStrategyForSession_l(sessionId)); 6759 chainCreated = true; 6760 } 6761 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6762 6763 if (chain->getEffectFromId_l(effect->id()) != 0) { 6764 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6765 this, effect->desc().name, chain.get()); 6766 return BAD_VALUE; 6767 } 6768 6769 status_t status = chain->addEffect_l(effect); 6770 if (status != NO_ERROR) { 6771 if (chainCreated) { 6772 removeEffectChain_l(chain); 6773 } 6774 return status; 6775 } 6776 6777 effect->setDevice(mDevice); 6778 effect->setMode(mAudioFlinger->getMode()); 6779 return NO_ERROR; 6780} 6781 6782void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6783 6784 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6785 effect_descriptor_t desc = effect->desc(); 6786 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6787 detachAuxEffect_l(effect->id()); 6788 } 6789 6790 sp<EffectChain> chain = effect->chain().promote(); 6791 if (chain != 0) { 6792 // remove effect chain if removing last effect 6793 if (chain->removeEffect_l(effect) == 0) { 6794 removeEffectChain_l(chain); 6795 } 6796 } else { 6797 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6798 } 6799} 6800 6801void AudioFlinger::ThreadBase::lockEffectChains_l( 6802 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6803{ 6804 effectChains = mEffectChains; 6805 for (size_t i = 0; i < mEffectChains.size(); i++) { 6806 mEffectChains[i]->lock(); 6807 } 6808} 6809 6810void AudioFlinger::ThreadBase::unlockEffectChains( 6811 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6812{ 6813 for (size_t i = 0; i < effectChains.size(); i++) { 6814 effectChains[i]->unlock(); 6815 } 6816} 6817 6818sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6819{ 6820 Mutex::Autolock _l(mLock); 6821 return getEffectChain_l(sessionId); 6822} 6823 6824sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6825{ 6826 size_t size = mEffectChains.size(); 6827 for (size_t i = 0; i < size; i++) { 6828 if (mEffectChains[i]->sessionId() == sessionId) { 6829 return mEffectChains[i]; 6830 } 6831 } 6832 return 0; 6833} 6834 6835void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6836{ 6837 Mutex::Autolock _l(mLock); 6838 size_t size = mEffectChains.size(); 6839 for (size_t i = 0; i < size; i++) { 6840 mEffectChains[i]->setMode_l(mode); 6841 } 6842} 6843 6844void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6845 const wp<EffectHandle>& handle, 6846 bool unpinIfLast) { 6847 6848 Mutex::Autolock _l(mLock); 6849 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6850 // delete the effect module if removing last handle on it 6851 if (effect->removeHandle(handle) == 0) { 6852 if (!effect->isPinned() || unpinIfLast) { 6853 removeEffect_l(effect); 6854 AudioSystem::unregisterEffect(effect->id()); 6855 } 6856 } 6857} 6858 6859status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6860{ 6861 int session = chain->sessionId(); 6862 int16_t *buffer = mMixBuffer; 6863 bool ownsBuffer = false; 6864 6865 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6866 if (session > 0) { 6867 // Only one effect chain can be present in direct output thread and it uses 6868 // the mix buffer as input 6869 if (mType != DIRECT) { 6870 size_t numSamples = mFrameCount * mChannelCount; 6871 buffer = new int16_t[numSamples]; 6872 memset(buffer, 0, numSamples * sizeof(int16_t)); 6873 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6874 ownsBuffer = true; 6875 } 6876 6877 // Attach all tracks with same session ID to this chain. 6878 for (size_t i = 0; i < mTracks.size(); ++i) { 6879 sp<Track> track = mTracks[i]; 6880 if (session == track->sessionId()) { 6881 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6882 track->setMainBuffer(buffer); 6883 chain->incTrackCnt(); 6884 } 6885 } 6886 6887 // indicate all active tracks in the chain 6888 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6889 sp<Track> track = mActiveTracks[i].promote(); 6890 if (track == 0) continue; 6891 if (session == track->sessionId()) { 6892 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6893 chain->incActiveTrackCnt(); 6894 } 6895 } 6896 } 6897 6898 chain->setInBuffer(buffer, ownsBuffer); 6899 chain->setOutBuffer(mMixBuffer); 6900 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6901 // chains list in order to be processed last as it contains output stage effects 6902 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6903 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6904 // after track specific effects and before output stage 6905 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6906 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6907 // Effect chain for other sessions are inserted at beginning of effect 6908 // chains list to be processed before output mix effects. Relative order between other 6909 // sessions is not important 6910 size_t size = mEffectChains.size(); 6911 size_t i = 0; 6912 for (i = 0; i < size; i++) { 6913 if (mEffectChains[i]->sessionId() < session) break; 6914 } 6915 mEffectChains.insertAt(chain, i); 6916 checkSuspendOnAddEffectChain_l(chain); 6917 6918 return NO_ERROR; 6919} 6920 6921size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6922{ 6923 int session = chain->sessionId(); 6924 6925 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6926 6927 for (size_t i = 0; i < mEffectChains.size(); i++) { 6928 if (chain == mEffectChains[i]) { 6929 mEffectChains.removeAt(i); 6930 // detach all active tracks from the chain 6931 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6932 sp<Track> track = mActiveTracks[i].promote(); 6933 if (track == 0) continue; 6934 if (session == track->sessionId()) { 6935 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6936 chain.get(), session); 6937 chain->decActiveTrackCnt(); 6938 } 6939 } 6940 6941 // detach all tracks with same session ID from this chain 6942 for (size_t i = 0; i < mTracks.size(); ++i) { 6943 sp<Track> track = mTracks[i]; 6944 if (session == track->sessionId()) { 6945 track->setMainBuffer(mMixBuffer); 6946 chain->decTrackCnt(); 6947 } 6948 } 6949 break; 6950 } 6951 } 6952 return mEffectChains.size(); 6953} 6954 6955status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6956 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6957{ 6958 Mutex::Autolock _l(mLock); 6959 return attachAuxEffect_l(track, EffectId); 6960} 6961 6962status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6963 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6964{ 6965 status_t status = NO_ERROR; 6966 6967 if (EffectId == 0) { 6968 track->setAuxBuffer(0, NULL); 6969 } else { 6970 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6971 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6972 if (effect != 0) { 6973 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6974 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6975 } else { 6976 status = INVALID_OPERATION; 6977 } 6978 } else { 6979 status = BAD_VALUE; 6980 } 6981 } 6982 return status; 6983} 6984 6985void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6986{ 6987 for (size_t i = 0; i < mTracks.size(); ++i) { 6988 sp<Track> track = mTracks[i]; 6989 if (track->auxEffectId() == effectId) { 6990 attachAuxEffect_l(track, 0); 6991 } 6992 } 6993} 6994 6995status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6996{ 6997 // only one chain per input thread 6998 if (mEffectChains.size() != 0) { 6999 return INVALID_OPERATION; 7000 } 7001 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7002 7003 chain->setInBuffer(NULL); 7004 chain->setOutBuffer(NULL); 7005 7006 checkSuspendOnAddEffectChain_l(chain); 7007 7008 mEffectChains.add(chain); 7009 7010 return NO_ERROR; 7011} 7012 7013size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7014{ 7015 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7016 ALOGW_IF(mEffectChains.size() != 1, 7017 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7018 chain.get(), mEffectChains.size(), this); 7019 if (mEffectChains.size() == 1) { 7020 mEffectChains.removeAt(0); 7021 } 7022 return 0; 7023} 7024 7025// ---------------------------------------------------------------------------- 7026// EffectModule implementation 7027// ---------------------------------------------------------------------------- 7028 7029#undef LOG_TAG 7030#define LOG_TAG "AudioFlinger::EffectModule" 7031 7032AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7033 const wp<AudioFlinger::EffectChain>& chain, 7034 effect_descriptor_t *desc, 7035 int id, 7036 int sessionId) 7037 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7038 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7039{ 7040 ALOGV("Constructor %p", this); 7041 int lStatus; 7042 if (thread == NULL) { 7043 return; 7044 } 7045 7046 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7047 7048 // create effect engine from effect factory 7049 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7050 7051 if (mStatus != NO_ERROR) { 7052 return; 7053 } 7054 lStatus = init(); 7055 if (lStatus < 0) { 7056 mStatus = lStatus; 7057 goto Error; 7058 } 7059 7060 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7061 mPinned = true; 7062 } 7063 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7064 return; 7065Error: 7066 EffectRelease(mEffectInterface); 7067 mEffectInterface = NULL; 7068 ALOGV("Constructor Error %d", mStatus); 7069} 7070 7071AudioFlinger::EffectModule::~EffectModule() 7072{ 7073 ALOGV("Destructor %p", this); 7074 if (mEffectInterface != NULL) { 7075 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7076 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7077 sp<ThreadBase> thread = mThread.promote(); 7078 if (thread != 0) { 7079 audio_stream_t *stream = thread->stream(); 7080 if (stream != NULL) { 7081 stream->remove_audio_effect(stream, mEffectInterface); 7082 } 7083 } 7084 } 7085 // release effect engine 7086 EffectRelease(mEffectInterface); 7087 } 7088} 7089 7090status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7091{ 7092 status_t status; 7093 7094 Mutex::Autolock _l(mLock); 7095 int priority = handle->priority(); 7096 size_t size = mHandles.size(); 7097 sp<EffectHandle> h; 7098 size_t i; 7099 for (i = 0; i < size; i++) { 7100 h = mHandles[i].promote(); 7101 if (h == 0) continue; 7102 if (h->priority() <= priority) break; 7103 } 7104 // if inserted in first place, move effect control from previous owner to this handle 7105 if (i == 0) { 7106 bool enabled = false; 7107 if (h != 0) { 7108 enabled = h->enabled(); 7109 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7110 } 7111 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7112 status = NO_ERROR; 7113 } else { 7114 status = ALREADY_EXISTS; 7115 } 7116 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7117 mHandles.insertAt(handle, i); 7118 return status; 7119} 7120 7121size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7122{ 7123 Mutex::Autolock _l(mLock); 7124 size_t size = mHandles.size(); 7125 size_t i; 7126 for (i = 0; i < size; i++) { 7127 if (mHandles[i] == handle) break; 7128 } 7129 if (i == size) { 7130 return size; 7131 } 7132 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7133 7134 bool enabled = false; 7135 EffectHandle *hdl = handle.unsafe_get(); 7136 if (hdl != NULL) { 7137 ALOGV("removeHandle() unsafe_get OK"); 7138 enabled = hdl->enabled(); 7139 } 7140 mHandles.removeAt(i); 7141 size = mHandles.size(); 7142 // if removed from first place, move effect control from this handle to next in line 7143 if (i == 0 && size != 0) { 7144 sp<EffectHandle> h = mHandles[0].promote(); 7145 if (h != 0) { 7146 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7147 } 7148 } 7149 7150 // Prevent calls to process() and other functions on effect interface from now on. 7151 // The effect engine will be released by the destructor when the last strong reference on 7152 // this object is released which can happen after next process is called. 7153 if (size == 0 && !mPinned) { 7154 mState = DESTROYED; 7155 } 7156 7157 return size; 7158} 7159 7160sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7161{ 7162 Mutex::Autolock _l(mLock); 7163 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7164} 7165 7166void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7167{ 7168 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7169 // keep a strong reference on this EffectModule to avoid calling the 7170 // destructor before we exit 7171 sp<EffectModule> keep(this); 7172 { 7173 sp<ThreadBase> thread = mThread.promote(); 7174 if (thread != 0) { 7175 thread->disconnectEffect(keep, handle, unpinIfLast); 7176 } 7177 } 7178} 7179 7180void AudioFlinger::EffectModule::updateState() { 7181 Mutex::Autolock _l(mLock); 7182 7183 switch (mState) { 7184 case RESTART: 7185 reset_l(); 7186 // FALL THROUGH 7187 7188 case STARTING: 7189 // clear auxiliary effect input buffer for next accumulation 7190 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7191 memset(mConfig.inputCfg.buffer.raw, 7192 0, 7193 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7194 } 7195 start_l(); 7196 mState = ACTIVE; 7197 break; 7198 case STOPPING: 7199 stop_l(); 7200 mDisableWaitCnt = mMaxDisableWaitCnt; 7201 mState = STOPPED; 7202 break; 7203 case STOPPED: 7204 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7205 // turn off sequence. 7206 if (--mDisableWaitCnt == 0) { 7207 reset_l(); 7208 mState = IDLE; 7209 } 7210 break; 7211 default: //IDLE , ACTIVE, DESTROYED 7212 break; 7213 } 7214} 7215 7216void AudioFlinger::EffectModule::process() 7217{ 7218 Mutex::Autolock _l(mLock); 7219 7220 if (mState == DESTROYED || mEffectInterface == NULL || 7221 mConfig.inputCfg.buffer.raw == NULL || 7222 mConfig.outputCfg.buffer.raw == NULL) { 7223 return; 7224 } 7225 7226 if (isProcessEnabled()) { 7227 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7228 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7229 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7230 mConfig.inputCfg.buffer.s32, 7231 mConfig.inputCfg.buffer.frameCount/2); 7232 } 7233 7234 // do the actual processing in the effect engine 7235 int ret = (*mEffectInterface)->process(mEffectInterface, 7236 &mConfig.inputCfg.buffer, 7237 &mConfig.outputCfg.buffer); 7238 7239 // force transition to IDLE state when engine is ready 7240 if (mState == STOPPED && ret == -ENODATA) { 7241 mDisableWaitCnt = 1; 7242 } 7243 7244 // clear auxiliary effect input buffer for next accumulation 7245 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7246 memset(mConfig.inputCfg.buffer.raw, 0, 7247 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7248 } 7249 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7250 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7251 // If an insert effect is idle and input buffer is different from output buffer, 7252 // accumulate input onto output 7253 sp<EffectChain> chain = mChain.promote(); 7254 if (chain != 0 && chain->activeTrackCnt() != 0) { 7255 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7256 int16_t *in = mConfig.inputCfg.buffer.s16; 7257 int16_t *out = mConfig.outputCfg.buffer.s16; 7258 for (size_t i = 0; i < frameCnt; i++) { 7259 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7260 } 7261 } 7262 } 7263} 7264 7265void AudioFlinger::EffectModule::reset_l() 7266{ 7267 if (mEffectInterface == NULL) { 7268 return; 7269 } 7270 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7271} 7272 7273status_t AudioFlinger::EffectModule::configure() 7274{ 7275 uint32_t channels; 7276 if (mEffectInterface == NULL) { 7277 return NO_INIT; 7278 } 7279 7280 sp<ThreadBase> thread = mThread.promote(); 7281 if (thread == 0) { 7282 return DEAD_OBJECT; 7283 } 7284 7285 // TODO: handle configuration of effects replacing track process 7286 if (thread->channelCount() == 1) { 7287 channels = AUDIO_CHANNEL_OUT_MONO; 7288 } else { 7289 channels = AUDIO_CHANNEL_OUT_STEREO; 7290 } 7291 7292 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7293 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7294 } else { 7295 mConfig.inputCfg.channels = channels; 7296 } 7297 mConfig.outputCfg.channels = channels; 7298 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7299 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7300 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7301 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7302 mConfig.inputCfg.bufferProvider.cookie = NULL; 7303 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7304 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7305 mConfig.outputCfg.bufferProvider.cookie = NULL; 7306 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7307 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7308 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7309 // Insert effect: 7310 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7311 // always overwrites output buffer: input buffer == output buffer 7312 // - in other sessions: 7313 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7314 // other effect: overwrites output buffer: input buffer == output buffer 7315 // Auxiliary effect: 7316 // accumulates in output buffer: input buffer != output buffer 7317 // Therefore: accumulate <=> input buffer != output buffer 7318 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7319 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7320 } else { 7321 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7322 } 7323 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7324 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7325 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7326 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7327 7328 ALOGV("configure() %p thread %p buffer %p framecount %d", 7329 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7330 7331 status_t cmdStatus; 7332 uint32_t size = sizeof(int); 7333 status_t status = (*mEffectInterface)->command(mEffectInterface, 7334 EFFECT_CMD_SET_CONFIG, 7335 sizeof(effect_config_t), 7336 &mConfig, 7337 &size, 7338 &cmdStatus); 7339 if (status == 0) { 7340 status = cmdStatus; 7341 } 7342 7343 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7344 (1000 * mConfig.outputCfg.buffer.frameCount); 7345 7346 return status; 7347} 7348 7349status_t AudioFlinger::EffectModule::init() 7350{ 7351 Mutex::Autolock _l(mLock); 7352 if (mEffectInterface == NULL) { 7353 return NO_INIT; 7354 } 7355 status_t cmdStatus; 7356 uint32_t size = sizeof(status_t); 7357 status_t status = (*mEffectInterface)->command(mEffectInterface, 7358 EFFECT_CMD_INIT, 7359 0, 7360 NULL, 7361 &size, 7362 &cmdStatus); 7363 if (status == 0) { 7364 status = cmdStatus; 7365 } 7366 return status; 7367} 7368 7369status_t AudioFlinger::EffectModule::start() 7370{ 7371 Mutex::Autolock _l(mLock); 7372 return start_l(); 7373} 7374 7375status_t AudioFlinger::EffectModule::start_l() 7376{ 7377 if (mEffectInterface == NULL) { 7378 return NO_INIT; 7379 } 7380 status_t cmdStatus; 7381 uint32_t size = sizeof(status_t); 7382 status_t status = (*mEffectInterface)->command(mEffectInterface, 7383 EFFECT_CMD_ENABLE, 7384 0, 7385 NULL, 7386 &size, 7387 &cmdStatus); 7388 if (status == 0) { 7389 status = cmdStatus; 7390 } 7391 if (status == 0 && 7392 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7393 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7394 sp<ThreadBase> thread = mThread.promote(); 7395 if (thread != 0) { 7396 audio_stream_t *stream = thread->stream(); 7397 if (stream != NULL) { 7398 stream->add_audio_effect(stream, mEffectInterface); 7399 } 7400 } 7401 } 7402 return status; 7403} 7404 7405status_t AudioFlinger::EffectModule::stop() 7406{ 7407 Mutex::Autolock _l(mLock); 7408 return stop_l(); 7409} 7410 7411status_t AudioFlinger::EffectModule::stop_l() 7412{ 7413 if (mEffectInterface == NULL) { 7414 return NO_INIT; 7415 } 7416 status_t cmdStatus; 7417 uint32_t size = sizeof(status_t); 7418 status_t status = (*mEffectInterface)->command(mEffectInterface, 7419 EFFECT_CMD_DISABLE, 7420 0, 7421 NULL, 7422 &size, 7423 &cmdStatus); 7424 if (status == 0) { 7425 status = cmdStatus; 7426 } 7427 if (status == 0 && 7428 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7429 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7430 sp<ThreadBase> thread = mThread.promote(); 7431 if (thread != 0) { 7432 audio_stream_t *stream = thread->stream(); 7433 if (stream != NULL) { 7434 stream->remove_audio_effect(stream, mEffectInterface); 7435 } 7436 } 7437 } 7438 return status; 7439} 7440 7441status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7442 uint32_t cmdSize, 7443 void *pCmdData, 7444 uint32_t *replySize, 7445 void *pReplyData) 7446{ 7447 Mutex::Autolock _l(mLock); 7448// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7449 7450 if (mState == DESTROYED || mEffectInterface == NULL) { 7451 return NO_INIT; 7452 } 7453 status_t status = (*mEffectInterface)->command(mEffectInterface, 7454 cmdCode, 7455 cmdSize, 7456 pCmdData, 7457 replySize, 7458 pReplyData); 7459 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7460 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7461 for (size_t i = 1; i < mHandles.size(); i++) { 7462 sp<EffectHandle> h = mHandles[i].promote(); 7463 if (h != 0) { 7464 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7465 } 7466 } 7467 } 7468 return status; 7469} 7470 7471status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7472{ 7473 7474 Mutex::Autolock _l(mLock); 7475 ALOGV("setEnabled %p enabled %d", this, enabled); 7476 7477 if (enabled != isEnabled()) { 7478 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7479 if (enabled && status != NO_ERROR) { 7480 return status; 7481 } 7482 7483 switch (mState) { 7484 // going from disabled to enabled 7485 case IDLE: 7486 mState = STARTING; 7487 break; 7488 case STOPPED: 7489 mState = RESTART; 7490 break; 7491 case STOPPING: 7492 mState = ACTIVE; 7493 break; 7494 7495 // going from enabled to disabled 7496 case RESTART: 7497 mState = STOPPED; 7498 break; 7499 case STARTING: 7500 mState = IDLE; 7501 break; 7502 case ACTIVE: 7503 mState = STOPPING; 7504 break; 7505 case DESTROYED: 7506 return NO_ERROR; // simply ignore as we are being destroyed 7507 } 7508 for (size_t i = 1; i < mHandles.size(); i++) { 7509 sp<EffectHandle> h = mHandles[i].promote(); 7510 if (h != 0) { 7511 h->setEnabled(enabled); 7512 } 7513 } 7514 } 7515 return NO_ERROR; 7516} 7517 7518bool AudioFlinger::EffectModule::isEnabled() const 7519{ 7520 switch (mState) { 7521 case RESTART: 7522 case STARTING: 7523 case ACTIVE: 7524 return true; 7525 case IDLE: 7526 case STOPPING: 7527 case STOPPED: 7528 case DESTROYED: 7529 default: 7530 return false; 7531 } 7532} 7533 7534bool AudioFlinger::EffectModule::isProcessEnabled() const 7535{ 7536 switch (mState) { 7537 case RESTART: 7538 case ACTIVE: 7539 case STOPPING: 7540 case STOPPED: 7541 return true; 7542 case IDLE: 7543 case STARTING: 7544 case DESTROYED: 7545 default: 7546 return false; 7547 } 7548} 7549 7550status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7551{ 7552 Mutex::Autolock _l(mLock); 7553 status_t status = NO_ERROR; 7554 7555 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7556 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7557 if (isProcessEnabled() && 7558 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7559 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7560 status_t cmdStatus; 7561 uint32_t volume[2]; 7562 uint32_t *pVolume = NULL; 7563 uint32_t size = sizeof(volume); 7564 volume[0] = *left; 7565 volume[1] = *right; 7566 if (controller) { 7567 pVolume = volume; 7568 } 7569 status = (*mEffectInterface)->command(mEffectInterface, 7570 EFFECT_CMD_SET_VOLUME, 7571 size, 7572 volume, 7573 &size, 7574 pVolume); 7575 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7576 *left = volume[0]; 7577 *right = volume[1]; 7578 } 7579 } 7580 return status; 7581} 7582 7583status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7584{ 7585 Mutex::Autolock _l(mLock); 7586 status_t status = NO_ERROR; 7587 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7588 // audio pre processing modules on RecordThread can receive both output and 7589 // input device indication in the same call 7590 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7591 if (dev) { 7592 status_t cmdStatus; 7593 uint32_t size = sizeof(status_t); 7594 7595 status = (*mEffectInterface)->command(mEffectInterface, 7596 EFFECT_CMD_SET_DEVICE, 7597 sizeof(uint32_t), 7598 &dev, 7599 &size, 7600 &cmdStatus); 7601 if (status == NO_ERROR) { 7602 status = cmdStatus; 7603 } 7604 } 7605 dev = device & AUDIO_DEVICE_IN_ALL; 7606 if (dev) { 7607 status_t cmdStatus; 7608 uint32_t size = sizeof(status_t); 7609 7610 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7611 EFFECT_CMD_SET_INPUT_DEVICE, 7612 sizeof(uint32_t), 7613 &dev, 7614 &size, 7615 &cmdStatus); 7616 if (status2 == NO_ERROR) { 7617 status2 = cmdStatus; 7618 } 7619 if (status == NO_ERROR) { 7620 status = status2; 7621 } 7622 } 7623 } 7624 return status; 7625} 7626 7627status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7628{ 7629 Mutex::Autolock _l(mLock); 7630 status_t status = NO_ERROR; 7631 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7632 status_t cmdStatus; 7633 uint32_t size = sizeof(status_t); 7634 status = (*mEffectInterface)->command(mEffectInterface, 7635 EFFECT_CMD_SET_AUDIO_MODE, 7636 sizeof(audio_mode_t), 7637 &mode, 7638 &size, 7639 &cmdStatus); 7640 if (status == NO_ERROR) { 7641 status = cmdStatus; 7642 } 7643 } 7644 return status; 7645} 7646 7647void AudioFlinger::EffectModule::setSuspended(bool suspended) 7648{ 7649 Mutex::Autolock _l(mLock); 7650 mSuspended = suspended; 7651} 7652 7653bool AudioFlinger::EffectModule::suspended() const 7654{ 7655 Mutex::Autolock _l(mLock); 7656 return mSuspended; 7657} 7658 7659status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7660{ 7661 const size_t SIZE = 256; 7662 char buffer[SIZE]; 7663 String8 result; 7664 7665 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7666 result.append(buffer); 7667 7668 bool locked = tryLock(mLock); 7669 // failed to lock - AudioFlinger is probably deadlocked 7670 if (!locked) { 7671 result.append("\t\tCould not lock Fx mutex:\n"); 7672 } 7673 7674 result.append("\t\tSession Status State Engine:\n"); 7675 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7676 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7677 result.append(buffer); 7678 7679 result.append("\t\tDescriptor:\n"); 7680 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7681 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7682 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7683 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7684 result.append(buffer); 7685 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7686 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7687 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7688 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7689 result.append(buffer); 7690 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7691 mDescriptor.apiVersion, 7692 mDescriptor.flags); 7693 result.append(buffer); 7694 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7695 mDescriptor.name); 7696 result.append(buffer); 7697 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7698 mDescriptor.implementor); 7699 result.append(buffer); 7700 7701 result.append("\t\t- Input configuration:\n"); 7702 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7703 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7704 (uint32_t)mConfig.inputCfg.buffer.raw, 7705 mConfig.inputCfg.buffer.frameCount, 7706 mConfig.inputCfg.samplingRate, 7707 mConfig.inputCfg.channels, 7708 mConfig.inputCfg.format); 7709 result.append(buffer); 7710 7711 result.append("\t\t- Output configuration:\n"); 7712 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7713 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7714 (uint32_t)mConfig.outputCfg.buffer.raw, 7715 mConfig.outputCfg.buffer.frameCount, 7716 mConfig.outputCfg.samplingRate, 7717 mConfig.outputCfg.channels, 7718 mConfig.outputCfg.format); 7719 result.append(buffer); 7720 7721 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7722 result.append(buffer); 7723 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7724 for (size_t i = 0; i < mHandles.size(); ++i) { 7725 sp<EffectHandle> handle = mHandles[i].promote(); 7726 if (handle != 0) { 7727 handle->dump(buffer, SIZE); 7728 result.append(buffer); 7729 } 7730 } 7731 7732 result.append("\n"); 7733 7734 write(fd, result.string(), result.length()); 7735 7736 if (locked) { 7737 mLock.unlock(); 7738 } 7739 7740 return NO_ERROR; 7741} 7742 7743// ---------------------------------------------------------------------------- 7744// EffectHandle implementation 7745// ---------------------------------------------------------------------------- 7746 7747#undef LOG_TAG 7748#define LOG_TAG "AudioFlinger::EffectHandle" 7749 7750AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7751 const sp<AudioFlinger::Client>& client, 7752 const sp<IEffectClient>& effectClient, 7753 int32_t priority) 7754 : BnEffect(), 7755 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7756 mPriority(priority), mHasControl(false), mEnabled(false) 7757{ 7758 ALOGV("constructor %p", this); 7759 7760 if (client == 0) { 7761 return; 7762 } 7763 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7764 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7765 if (mCblkMemory != 0) { 7766 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7767 7768 if (mCblk != NULL) { 7769 new(mCblk) effect_param_cblk_t(); 7770 mBuffer = (uint8_t *)mCblk + bufOffset; 7771 } 7772 } else { 7773 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7774 return; 7775 } 7776} 7777 7778AudioFlinger::EffectHandle::~EffectHandle() 7779{ 7780 ALOGV("Destructor %p", this); 7781 disconnect(false); 7782 ALOGV("Destructor DONE %p", this); 7783} 7784 7785status_t AudioFlinger::EffectHandle::enable() 7786{ 7787 ALOGV("enable %p", this); 7788 if (!mHasControl) return INVALID_OPERATION; 7789 if (mEffect == 0) return DEAD_OBJECT; 7790 7791 if (mEnabled) { 7792 return NO_ERROR; 7793 } 7794 7795 mEnabled = true; 7796 7797 sp<ThreadBase> thread = mEffect->thread().promote(); 7798 if (thread != 0) { 7799 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7800 } 7801 7802 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7803 if (mEffect->suspended()) { 7804 return NO_ERROR; 7805 } 7806 7807 status_t status = mEffect->setEnabled(true); 7808 if (status != NO_ERROR) { 7809 if (thread != 0) { 7810 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7811 } 7812 mEnabled = false; 7813 } 7814 return status; 7815} 7816 7817status_t AudioFlinger::EffectHandle::disable() 7818{ 7819 ALOGV("disable %p", this); 7820 if (!mHasControl) return INVALID_OPERATION; 7821 if (mEffect == 0) return DEAD_OBJECT; 7822 7823 if (!mEnabled) { 7824 return NO_ERROR; 7825 } 7826 mEnabled = false; 7827 7828 if (mEffect->suspended()) { 7829 return NO_ERROR; 7830 } 7831 7832 status_t status = mEffect->setEnabled(false); 7833 7834 sp<ThreadBase> thread = mEffect->thread().promote(); 7835 if (thread != 0) { 7836 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7837 } 7838 7839 return status; 7840} 7841 7842void AudioFlinger::EffectHandle::disconnect() 7843{ 7844 disconnect(true); 7845} 7846 7847void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7848{ 7849 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7850 if (mEffect == 0) { 7851 return; 7852 } 7853 mEffect->disconnect(this, unpinIfLast); 7854 7855 if (mHasControl && mEnabled) { 7856 sp<ThreadBase> thread = mEffect->thread().promote(); 7857 if (thread != 0) { 7858 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7859 } 7860 } 7861 7862 // release sp on module => module destructor can be called now 7863 mEffect.clear(); 7864 if (mClient != 0) { 7865 if (mCblk != NULL) { 7866 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7867 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7868 } 7869 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7870 // Client destructor must run with AudioFlinger mutex locked 7871 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7872 mClient.clear(); 7873 } 7874} 7875 7876status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7877 uint32_t cmdSize, 7878 void *pCmdData, 7879 uint32_t *replySize, 7880 void *pReplyData) 7881{ 7882// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7883// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7884 7885 // only get parameter command is permitted for applications not controlling the effect 7886 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7887 return INVALID_OPERATION; 7888 } 7889 if (mEffect == 0) return DEAD_OBJECT; 7890 if (mClient == 0) return INVALID_OPERATION; 7891 7892 // handle commands that are not forwarded transparently to effect engine 7893 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7894 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7895 // no risk to block the whole media server process or mixer threads is we are stuck here 7896 Mutex::Autolock _l(mCblk->lock); 7897 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7898 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7899 mCblk->serverIndex = 0; 7900 mCblk->clientIndex = 0; 7901 return BAD_VALUE; 7902 } 7903 status_t status = NO_ERROR; 7904 while (mCblk->serverIndex < mCblk->clientIndex) { 7905 int reply; 7906 uint32_t rsize = sizeof(int); 7907 int *p = (int *)(mBuffer + mCblk->serverIndex); 7908 int size = *p++; 7909 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7910 ALOGW("command(): invalid parameter block size"); 7911 break; 7912 } 7913 effect_param_t *param = (effect_param_t *)p; 7914 if (param->psize == 0 || param->vsize == 0) { 7915 ALOGW("command(): null parameter or value size"); 7916 mCblk->serverIndex += size; 7917 continue; 7918 } 7919 uint32_t psize = sizeof(effect_param_t) + 7920 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7921 param->vsize; 7922 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7923 psize, 7924 p, 7925 &rsize, 7926 &reply); 7927 // stop at first error encountered 7928 if (ret != NO_ERROR) { 7929 status = ret; 7930 *(int *)pReplyData = reply; 7931 break; 7932 } else if (reply != NO_ERROR) { 7933 *(int *)pReplyData = reply; 7934 break; 7935 } 7936 mCblk->serverIndex += size; 7937 } 7938 mCblk->serverIndex = 0; 7939 mCblk->clientIndex = 0; 7940 return status; 7941 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7942 *(int *)pReplyData = NO_ERROR; 7943 return enable(); 7944 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7945 *(int *)pReplyData = NO_ERROR; 7946 return disable(); 7947 } 7948 7949 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7950} 7951 7952void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7953{ 7954 ALOGV("setControl %p control %d", this, hasControl); 7955 7956 mHasControl = hasControl; 7957 mEnabled = enabled; 7958 7959 if (signal && mEffectClient != 0) { 7960 mEffectClient->controlStatusChanged(hasControl); 7961 } 7962} 7963 7964void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7965 uint32_t cmdSize, 7966 void *pCmdData, 7967 uint32_t replySize, 7968 void *pReplyData) 7969{ 7970 if (mEffectClient != 0) { 7971 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7972 } 7973} 7974 7975 7976 7977void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7978{ 7979 if (mEffectClient != 0) { 7980 mEffectClient->enableStatusChanged(enabled); 7981 } 7982} 7983 7984status_t AudioFlinger::EffectHandle::onTransact( 7985 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7986{ 7987 return BnEffect::onTransact(code, data, reply, flags); 7988} 7989 7990 7991void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7992{ 7993 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7994 7995 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7996 (mClient == 0) ? getpid_cached : mClient->pid(), 7997 mPriority, 7998 mHasControl, 7999 !locked, 8000 mCblk ? mCblk->clientIndex : 0, 8001 mCblk ? mCblk->serverIndex : 0 8002 ); 8003 8004 if (locked) { 8005 mCblk->lock.unlock(); 8006 } 8007} 8008 8009#undef LOG_TAG 8010#define LOG_TAG "AudioFlinger::EffectChain" 8011 8012AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8013 int sessionId) 8014 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8015 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8016 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8017{ 8018 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8019 if (thread == NULL) { 8020 return; 8021 } 8022 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8023 thread->frameCount(); 8024} 8025 8026AudioFlinger::EffectChain::~EffectChain() 8027{ 8028 if (mOwnInBuffer) { 8029 delete mInBuffer; 8030 } 8031 8032} 8033 8034// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8035sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8036{ 8037 size_t size = mEffects.size(); 8038 8039 for (size_t i = 0; i < size; i++) { 8040 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8041 return mEffects[i]; 8042 } 8043 } 8044 return 0; 8045} 8046 8047// getEffectFromId_l() must be called with ThreadBase::mLock held 8048sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8049{ 8050 size_t size = mEffects.size(); 8051 8052 for (size_t i = 0; i < size; i++) { 8053 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8054 if (id == 0 || mEffects[i]->id() == id) { 8055 return mEffects[i]; 8056 } 8057 } 8058 return 0; 8059} 8060 8061// getEffectFromType_l() must be called with ThreadBase::mLock held 8062sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8063 const effect_uuid_t *type) 8064{ 8065 size_t size = mEffects.size(); 8066 8067 for (size_t i = 0; i < size; i++) { 8068 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8069 return mEffects[i]; 8070 } 8071 } 8072 return 0; 8073} 8074 8075// Must be called with EffectChain::mLock locked 8076void AudioFlinger::EffectChain::process_l() 8077{ 8078 sp<ThreadBase> thread = mThread.promote(); 8079 if (thread == 0) { 8080 ALOGW("process_l(): cannot promote mixer thread"); 8081 return; 8082 } 8083 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8084 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8085 // always process effects unless no more tracks are on the session and the effect tail 8086 // has been rendered 8087 bool doProcess = true; 8088 if (!isGlobalSession) { 8089 bool tracksOnSession = (trackCnt() != 0); 8090 8091 if (!tracksOnSession && mTailBufferCount == 0) { 8092 doProcess = false; 8093 } 8094 8095 if (activeTrackCnt() == 0) { 8096 // if no track is active and the effect tail has not been rendered, 8097 // the input buffer must be cleared here as the mixer process will not do it 8098 if (tracksOnSession || mTailBufferCount > 0) { 8099 size_t numSamples = thread->frameCount() * thread->channelCount(); 8100 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8101 if (mTailBufferCount > 0) { 8102 mTailBufferCount--; 8103 } 8104 } 8105 } 8106 } 8107 8108 size_t size = mEffects.size(); 8109 if (doProcess) { 8110 for (size_t i = 0; i < size; i++) { 8111 mEffects[i]->process(); 8112 } 8113 } 8114 for (size_t i = 0; i < size; i++) { 8115 mEffects[i]->updateState(); 8116 } 8117} 8118 8119// addEffect_l() must be called with PlaybackThread::mLock held 8120status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8121{ 8122 effect_descriptor_t desc = effect->desc(); 8123 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8124 8125 Mutex::Autolock _l(mLock); 8126 effect->setChain(this); 8127 sp<ThreadBase> thread = mThread.promote(); 8128 if (thread == 0) { 8129 return NO_INIT; 8130 } 8131 effect->setThread(thread); 8132 8133 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8134 // Auxiliary effects are inserted at the beginning of mEffects vector as 8135 // they are processed first and accumulated in chain input buffer 8136 mEffects.insertAt(effect, 0); 8137 8138 // the input buffer for auxiliary effect contains mono samples in 8139 // 32 bit format. This is to avoid saturation in AudoMixer 8140 // accumulation stage. Saturation is done in EffectModule::process() before 8141 // calling the process in effect engine 8142 size_t numSamples = thread->frameCount(); 8143 int32_t *buffer = new int32_t[numSamples]; 8144 memset(buffer, 0, numSamples * sizeof(int32_t)); 8145 effect->setInBuffer((int16_t *)buffer); 8146 // auxiliary effects output samples to chain input buffer for further processing 8147 // by insert effects 8148 effect->setOutBuffer(mInBuffer); 8149 } else { 8150 // Insert effects are inserted at the end of mEffects vector as they are processed 8151 // after track and auxiliary effects. 8152 // Insert effect order as a function of indicated preference: 8153 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8154 // another effect is present 8155 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8156 // last effect claiming first position 8157 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8158 // first effect claiming last position 8159 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8160 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8161 // already present 8162 8163 size_t size = mEffects.size(); 8164 size_t idx_insert = size; 8165 ssize_t idx_insert_first = -1; 8166 ssize_t idx_insert_last = -1; 8167 8168 for (size_t i = 0; i < size; i++) { 8169 effect_descriptor_t d = mEffects[i]->desc(); 8170 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8171 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8172 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8173 // check invalid effect chaining combinations 8174 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8175 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8176 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8177 return INVALID_OPERATION; 8178 } 8179 // remember position of first insert effect and by default 8180 // select this as insert position for new effect 8181 if (idx_insert == size) { 8182 idx_insert = i; 8183 } 8184 // remember position of last insert effect claiming 8185 // first position 8186 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8187 idx_insert_first = i; 8188 } 8189 // remember position of first insert effect claiming 8190 // last position 8191 if (iPref == EFFECT_FLAG_INSERT_LAST && 8192 idx_insert_last == -1) { 8193 idx_insert_last = i; 8194 } 8195 } 8196 } 8197 8198 // modify idx_insert from first position if needed 8199 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8200 if (idx_insert_last != -1) { 8201 idx_insert = idx_insert_last; 8202 } else { 8203 idx_insert = size; 8204 } 8205 } else { 8206 if (idx_insert_first != -1) { 8207 idx_insert = idx_insert_first + 1; 8208 } 8209 } 8210 8211 // always read samples from chain input buffer 8212 effect->setInBuffer(mInBuffer); 8213 8214 // if last effect in the chain, output samples to chain 8215 // output buffer, otherwise to chain input buffer 8216 if (idx_insert == size) { 8217 if (idx_insert != 0) { 8218 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8219 mEffects[idx_insert-1]->configure(); 8220 } 8221 effect->setOutBuffer(mOutBuffer); 8222 } else { 8223 effect->setOutBuffer(mInBuffer); 8224 } 8225 mEffects.insertAt(effect, idx_insert); 8226 8227 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8228 } 8229 effect->configure(); 8230 return NO_ERROR; 8231} 8232 8233// removeEffect_l() must be called with PlaybackThread::mLock held 8234size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8235{ 8236 Mutex::Autolock _l(mLock); 8237 size_t size = mEffects.size(); 8238 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8239 8240 for (size_t i = 0; i < size; i++) { 8241 if (effect == mEffects[i]) { 8242 // calling stop here will remove pre-processing effect from the audio HAL. 8243 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8244 // the middle of a read from audio HAL 8245 if (mEffects[i]->state() == EffectModule::ACTIVE || 8246 mEffects[i]->state() == EffectModule::STOPPING) { 8247 mEffects[i]->stop(); 8248 } 8249 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8250 delete[] effect->inBuffer(); 8251 } else { 8252 if (i == size - 1 && i != 0) { 8253 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8254 mEffects[i - 1]->configure(); 8255 } 8256 } 8257 mEffects.removeAt(i); 8258 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8259 break; 8260 } 8261 } 8262 8263 return mEffects.size(); 8264} 8265 8266// setDevice_l() must be called with PlaybackThread::mLock held 8267void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8268{ 8269 size_t size = mEffects.size(); 8270 for (size_t i = 0; i < size; i++) { 8271 mEffects[i]->setDevice(device); 8272 } 8273} 8274 8275// setMode_l() must be called with PlaybackThread::mLock held 8276void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8277{ 8278 size_t size = mEffects.size(); 8279 for (size_t i = 0; i < size; i++) { 8280 mEffects[i]->setMode(mode); 8281 } 8282} 8283 8284// setVolume_l() must be called with PlaybackThread::mLock held 8285bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8286{ 8287 uint32_t newLeft = *left; 8288 uint32_t newRight = *right; 8289 bool hasControl = false; 8290 int ctrlIdx = -1; 8291 size_t size = mEffects.size(); 8292 8293 // first update volume controller 8294 for (size_t i = size; i > 0; i--) { 8295 if (mEffects[i - 1]->isProcessEnabled() && 8296 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8297 ctrlIdx = i - 1; 8298 hasControl = true; 8299 break; 8300 } 8301 } 8302 8303 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8304 if (hasControl) { 8305 *left = mNewLeftVolume; 8306 *right = mNewRightVolume; 8307 } 8308 return hasControl; 8309 } 8310 8311 mVolumeCtrlIdx = ctrlIdx; 8312 mLeftVolume = newLeft; 8313 mRightVolume = newRight; 8314 8315 // second get volume update from volume controller 8316 if (ctrlIdx >= 0) { 8317 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8318 mNewLeftVolume = newLeft; 8319 mNewRightVolume = newRight; 8320 } 8321 // then indicate volume to all other effects in chain. 8322 // Pass altered volume to effects before volume controller 8323 // and requested volume to effects after controller 8324 uint32_t lVol = newLeft; 8325 uint32_t rVol = newRight; 8326 8327 for (size_t i = 0; i < size; i++) { 8328 if ((int)i == ctrlIdx) continue; 8329 // this also works for ctrlIdx == -1 when there is no volume controller 8330 if ((int)i > ctrlIdx) { 8331 lVol = *left; 8332 rVol = *right; 8333 } 8334 mEffects[i]->setVolume(&lVol, &rVol, false); 8335 } 8336 *left = newLeft; 8337 *right = newRight; 8338 8339 return hasControl; 8340} 8341 8342status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8343{ 8344 const size_t SIZE = 256; 8345 char buffer[SIZE]; 8346 String8 result; 8347 8348 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8349 result.append(buffer); 8350 8351 bool locked = tryLock(mLock); 8352 // failed to lock - AudioFlinger is probably deadlocked 8353 if (!locked) { 8354 result.append("\tCould not lock mutex:\n"); 8355 } 8356 8357 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8358 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8359 mEffects.size(), 8360 (uint32_t)mInBuffer, 8361 (uint32_t)mOutBuffer, 8362 mActiveTrackCnt); 8363 result.append(buffer); 8364 write(fd, result.string(), result.size()); 8365 8366 for (size_t i = 0; i < mEffects.size(); ++i) { 8367 sp<EffectModule> effect = mEffects[i]; 8368 if (effect != 0) { 8369 effect->dump(fd, args); 8370 } 8371 } 8372 8373 if (locked) { 8374 mLock.unlock(); 8375 } 8376 8377 return NO_ERROR; 8378} 8379 8380// must be called with ThreadBase::mLock held 8381void AudioFlinger::EffectChain::setEffectSuspended_l( 8382 const effect_uuid_t *type, bool suspend) 8383{ 8384 sp<SuspendedEffectDesc> desc; 8385 // use effect type UUID timelow as key as there is no real risk of identical 8386 // timeLow fields among effect type UUIDs. 8387 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8388 if (suspend) { 8389 if (index >= 0) { 8390 desc = mSuspendedEffects.valueAt(index); 8391 } else { 8392 desc = new SuspendedEffectDesc(); 8393 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8394 mSuspendedEffects.add(type->timeLow, desc); 8395 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8396 } 8397 if (desc->mRefCount++ == 0) { 8398 sp<EffectModule> effect = getEffectIfEnabled(type); 8399 if (effect != 0) { 8400 desc->mEffect = effect; 8401 effect->setSuspended(true); 8402 effect->setEnabled(false); 8403 } 8404 } 8405 } else { 8406 if (index < 0) { 8407 return; 8408 } 8409 desc = mSuspendedEffects.valueAt(index); 8410 if (desc->mRefCount <= 0) { 8411 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8412 desc->mRefCount = 1; 8413 } 8414 if (--desc->mRefCount == 0) { 8415 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8416 if (desc->mEffect != 0) { 8417 sp<EffectModule> effect = desc->mEffect.promote(); 8418 if (effect != 0) { 8419 effect->setSuspended(false); 8420 sp<EffectHandle> handle = effect->controlHandle(); 8421 if (handle != 0) { 8422 effect->setEnabled(handle->enabled()); 8423 } 8424 } 8425 desc->mEffect.clear(); 8426 } 8427 mSuspendedEffects.removeItemsAt(index); 8428 } 8429 } 8430} 8431 8432// must be called with ThreadBase::mLock held 8433void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8434{ 8435 sp<SuspendedEffectDesc> desc; 8436 8437 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8438 if (suspend) { 8439 if (index >= 0) { 8440 desc = mSuspendedEffects.valueAt(index); 8441 } else { 8442 desc = new SuspendedEffectDesc(); 8443 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8444 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8445 } 8446 if (desc->mRefCount++ == 0) { 8447 Vector< sp<EffectModule> > effects; 8448 getSuspendEligibleEffects(effects); 8449 for (size_t i = 0; i < effects.size(); i++) { 8450 setEffectSuspended_l(&effects[i]->desc().type, true); 8451 } 8452 } 8453 } else { 8454 if (index < 0) { 8455 return; 8456 } 8457 desc = mSuspendedEffects.valueAt(index); 8458 if (desc->mRefCount <= 0) { 8459 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8460 desc->mRefCount = 1; 8461 } 8462 if (--desc->mRefCount == 0) { 8463 Vector<const effect_uuid_t *> types; 8464 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8465 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8466 continue; 8467 } 8468 types.add(&mSuspendedEffects.valueAt(i)->mType); 8469 } 8470 for (size_t i = 0; i < types.size(); i++) { 8471 setEffectSuspended_l(types[i], false); 8472 } 8473 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8474 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8475 } 8476 } 8477} 8478 8479 8480// The volume effect is used for automated tests only 8481#ifndef OPENSL_ES_H_ 8482static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8483 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8484const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8485#endif //OPENSL_ES_H_ 8486 8487bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8488{ 8489 // auxiliary effects and visualizer are never suspended on output mix 8490 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8491 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8492 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8493 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8494 return false; 8495 } 8496 return true; 8497} 8498 8499void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8500{ 8501 effects.clear(); 8502 for (size_t i = 0; i < mEffects.size(); i++) { 8503 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8504 effects.add(mEffects[i]); 8505 } 8506 } 8507} 8508 8509sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8510 const effect_uuid_t *type) 8511{ 8512 sp<EffectModule> effect = getEffectFromType_l(type); 8513 return effect != 0 && effect->isEnabled() ? effect : 0; 8514} 8515 8516void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8517 bool enabled) 8518{ 8519 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8520 if (enabled) { 8521 if (index < 0) { 8522 // if the effect is not suspend check if all effects are suspended 8523 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8524 if (index < 0) { 8525 return; 8526 } 8527 if (!isEffectEligibleForSuspend(effect->desc())) { 8528 return; 8529 } 8530 setEffectSuspended_l(&effect->desc().type, enabled); 8531 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8532 if (index < 0) { 8533 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8534 return; 8535 } 8536 } 8537 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8538 effect->desc().type.timeLow); 8539 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8540 // if effect is requested to suspended but was not yet enabled, supend it now. 8541 if (desc->mEffect == 0) { 8542 desc->mEffect = effect; 8543 effect->setEnabled(false); 8544 effect->setSuspended(true); 8545 } 8546 } else { 8547 if (index < 0) { 8548 return; 8549 } 8550 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8551 effect->desc().type.timeLow); 8552 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8553 desc->mEffect.clear(); 8554 effect->setSuspended(false); 8555 } 8556} 8557 8558#undef LOG_TAG 8559#define LOG_TAG "AudioFlinger" 8560 8561// ---------------------------------------------------------------------------- 8562 8563status_t AudioFlinger::onTransact( 8564 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8565{ 8566 return BnAudioFlinger::onTransact(code, data, reply, flags); 8567} 8568 8569}; // namespace android 8570