AudioFlinger.cpp revision 9bd23229fdec1657398abc682ccccfce1c95f8aa
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 return mAudioMixer->getTrackName(channelMask); 2668} 2669 2670// deleteTrackName_l() must be called with ThreadBase::mLock held 2671void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2672{ 2673 ALOGV("remove track (%d) and delete from mixer", name); 2674 mAudioMixer->deleteTrackName(name); 2675} 2676 2677// checkForNewParameters_l() must be called with ThreadBase::mLock held 2678bool AudioFlinger::MixerThread::checkForNewParameters_l() 2679{ 2680 bool reconfig = false; 2681 2682 while (!mNewParameters.isEmpty()) { 2683 status_t status = NO_ERROR; 2684 String8 keyValuePair = mNewParameters[0]; 2685 AudioParameter param = AudioParameter(keyValuePair); 2686 int value; 2687 2688 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2689 reconfig = true; 2690 } 2691 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2692 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2693 status = BAD_VALUE; 2694 } else { 2695 reconfig = true; 2696 } 2697 } 2698 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2699 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2700 status = BAD_VALUE; 2701 } else { 2702 reconfig = true; 2703 } 2704 } 2705 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2706 // do not accept frame count changes if tracks are open as the track buffer 2707 // size depends on frame count and correct behavior would not be guaranteed 2708 // if frame count is changed after track creation 2709 if (!mTracks.isEmpty()) { 2710 status = INVALID_OPERATION; 2711 } else { 2712 reconfig = true; 2713 } 2714 } 2715 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2716#ifdef ADD_BATTERY_DATA 2717 // when changing the audio output device, call addBatteryData to notify 2718 // the change 2719 if ((int)mDevice != value) { 2720 uint32_t params = 0; 2721 // check whether speaker is on 2722 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2723 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2724 } 2725 2726 int deviceWithoutSpeaker 2727 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2728 // check if any other device (except speaker) is on 2729 if (value & deviceWithoutSpeaker ) { 2730 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2731 } 2732 2733 if (params != 0) { 2734 addBatteryData(params); 2735 } 2736 } 2737#endif 2738 2739 // forward device change to effects that have requested to be 2740 // aware of attached audio device. 2741 mDevice = (uint32_t)value; 2742 for (size_t i = 0; i < mEffectChains.size(); i++) { 2743 mEffectChains[i]->setDevice_l(mDevice); 2744 } 2745 } 2746 2747 if (status == NO_ERROR) { 2748 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2749 keyValuePair.string()); 2750 if (!mStandby && status == INVALID_OPERATION) { 2751 mOutput->stream->common.standby(&mOutput->stream->common); 2752 mStandby = true; 2753 mBytesWritten = 0; 2754 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2755 keyValuePair.string()); 2756 } 2757 if (status == NO_ERROR && reconfig) { 2758 delete mAudioMixer; 2759 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2760 mAudioMixer = NULL; 2761 readOutputParameters(); 2762 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2763 for (size_t i = 0; i < mTracks.size() ; i++) { 2764 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 2765 if (name < 0) break; 2766 mTracks[i]->mName = name; 2767 // limit track sample rate to 2 x new output sample rate 2768 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2769 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2770 } 2771 } 2772 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2773 } 2774 } 2775 2776 mNewParameters.removeAt(0); 2777 2778 mParamStatus = status; 2779 mParamCond.signal(); 2780 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2781 // already timed out waiting for the status and will never signal the condition. 2782 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2783 } 2784 return reconfig; 2785} 2786 2787status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2788{ 2789 const size_t SIZE = 256; 2790 char buffer[SIZE]; 2791 String8 result; 2792 2793 PlaybackThread::dumpInternals(fd, args); 2794 2795 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2796 result.append(buffer); 2797 write(fd, result.string(), result.size()); 2798 return NO_ERROR; 2799} 2800 2801uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 2802{ 2803 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2804} 2805 2806uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 2807{ 2808 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2809} 2810 2811void AudioFlinger::MixerThread::cacheParameters_l() 2812{ 2813 PlaybackThread::cacheParameters_l(); 2814 2815 // FIXME: Relaxed timing because of a certain device that can't meet latency 2816 // Should be reduced to 2x after the vendor fixes the driver issue 2817 // increase threshold again due to low power audio mode. The way this warning 2818 // threshold is calculated and its usefulness should be reconsidered anyway. 2819 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2820} 2821 2822// ---------------------------------------------------------------------------- 2823AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2824 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2825 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2826 // mLeftVolFloat, mRightVolFloat 2827 // mLeftVolShort, mRightVolShort 2828{ 2829} 2830 2831AudioFlinger::DirectOutputThread::~DirectOutputThread() 2832{ 2833} 2834 2835AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2836 Vector< sp<Track> > *tracksToRemove 2837) 2838{ 2839 sp<Track> trackToRemove; 2840 2841 mixer_state mixerStatus = MIXER_IDLE; 2842 2843 // find out which tracks need to be processed 2844 if (mActiveTracks.size() != 0) { 2845 sp<Track> t = mActiveTracks[0].promote(); 2846 // The track died recently 2847 if (t == 0) return MIXER_IDLE; 2848 2849 Track* const track = t.get(); 2850 audio_track_cblk_t* cblk = track->cblk(); 2851 2852 // The first time a track is added we wait 2853 // for all its buffers to be filled before processing it 2854 if (cblk->framesReady() && track->isReady() && 2855 !track->isPaused() && !track->isTerminated()) 2856 { 2857 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2858 2859 if (track->mFillingUpStatus == Track::FS_FILLED) { 2860 track->mFillingUpStatus = Track::FS_ACTIVE; 2861 mLeftVolFloat = mRightVolFloat = 0; 2862 mLeftVolShort = mRightVolShort = 0; 2863 if (track->mState == TrackBase::RESUMING) { 2864 track->mState = TrackBase::ACTIVE; 2865 rampVolume = true; 2866 } 2867 } else if (cblk->server != 0) { 2868 // If the track is stopped before the first frame was mixed, 2869 // do not apply ramp 2870 rampVolume = true; 2871 } 2872 // compute volume for this track 2873 float left, right; 2874 if (track->isMuted() || mMasterMute || track->isPausing() || 2875 mStreamTypes[track->streamType()].mute) { 2876 left = right = 0; 2877 if (track->isPausing()) { 2878 track->setPaused(); 2879 } 2880 } else { 2881 float typeVolume = mStreamTypes[track->streamType()].volume; 2882 float v = mMasterVolume * typeVolume; 2883 uint32_t vlr = cblk->getVolumeLR(); 2884 float v_clamped = v * (vlr & 0xFFFF); 2885 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2886 left = v_clamped/MAX_GAIN; 2887 v_clamped = v * (vlr >> 16); 2888 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2889 right = v_clamped/MAX_GAIN; 2890 } 2891 2892 if (left != mLeftVolFloat || right != mRightVolFloat) { 2893 mLeftVolFloat = left; 2894 mRightVolFloat = right; 2895 2896 // If audio HAL implements volume control, 2897 // force software volume to nominal value 2898 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2899 left = 1.0f; 2900 right = 1.0f; 2901 } 2902 2903 // Convert volumes from float to 8.24 2904 uint32_t vl = (uint32_t)(left * (1 << 24)); 2905 uint32_t vr = (uint32_t)(right * (1 << 24)); 2906 2907 // Delegate volume control to effect in track effect chain if needed 2908 // only one effect chain can be present on DirectOutputThread, so if 2909 // there is one, the track is connected to it 2910 if (!mEffectChains.isEmpty()) { 2911 // Do not ramp volume if volume is controlled by effect 2912 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2913 rampVolume = false; 2914 } 2915 } 2916 2917 // Convert volumes from 8.24 to 4.12 format 2918 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2919 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2920 leftVol = (uint16_t)v_clamped; 2921 v_clamped = (vr + (1 << 11)) >> 12; 2922 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2923 rightVol = (uint16_t)v_clamped; 2924 } else { 2925 leftVol = mLeftVolShort; 2926 rightVol = mRightVolShort; 2927 rampVolume = false; 2928 } 2929 2930 // reset retry count 2931 track->mRetryCount = kMaxTrackRetriesDirect; 2932 mActiveTrack = t; 2933 mixerStatus = MIXER_TRACKS_READY; 2934 } else { 2935 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2936 if (track->isStopped()) { 2937 track->reset(); 2938 } 2939 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2940 // We have consumed all the buffers of this track. 2941 // Remove it from the list of active tracks. 2942 // TODO: implement behavior for compressed audio 2943 size_t audioHALFrames = 2944 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2945 size_t framesWritten = 2946 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2947 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2948 trackToRemove = track; 2949 } 2950 } else { 2951 // No buffers for this track. Give it a few chances to 2952 // fill a buffer, then remove it from active list. 2953 if (--(track->mRetryCount) <= 0) { 2954 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2955 trackToRemove = track; 2956 } else { 2957 mixerStatus = MIXER_TRACKS_ENABLED; 2958 } 2959 } 2960 } 2961 } 2962 2963 // FIXME merge this with similar code for removing multiple tracks 2964 // remove all the tracks that need to be... 2965 if (CC_UNLIKELY(trackToRemove != 0)) { 2966 tracksToRemove->add(trackToRemove); 2967 mActiveTracks.remove(trackToRemove); 2968 if (!mEffectChains.isEmpty()) { 2969 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2970 trackToRemove->sessionId()); 2971 mEffectChains[0]->decActiveTrackCnt(); 2972 } 2973 if (trackToRemove->isTerminated()) { 2974 removeTrack_l(trackToRemove); 2975 } 2976 } 2977 2978 return mixerStatus; 2979} 2980 2981void AudioFlinger::DirectOutputThread::threadLoop_mix() 2982{ 2983 AudioBufferProvider::Buffer buffer; 2984 size_t frameCount = mFrameCount; 2985 int8_t *curBuf = (int8_t *)mMixBuffer; 2986 // output audio to hardware 2987 while (frameCount) { 2988 buffer.frameCount = frameCount; 2989 mActiveTrack->getNextBuffer(&buffer); 2990 if (CC_UNLIKELY(buffer.raw == NULL)) { 2991 memset(curBuf, 0, frameCount * mFrameSize); 2992 break; 2993 } 2994 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2995 frameCount -= buffer.frameCount; 2996 curBuf += buffer.frameCount * mFrameSize; 2997 mActiveTrack->releaseBuffer(&buffer); 2998 } 2999 sleepTime = 0; 3000 standbyTime = systemTime() + standbyDelay; 3001 mActiveTrack.clear(); 3002 3003 // apply volume 3004 3005 // Do not apply volume on compressed audio 3006 if (!audio_is_linear_pcm(mFormat)) { 3007 return; 3008 } 3009 3010 // convert to signed 16 bit before volume calculation 3011 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3012 size_t count = mFrameCount * mChannelCount; 3013 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3014 int16_t *dst = mMixBuffer + count-1; 3015 while (count--) { 3016 *dst-- = (int16_t)(*src--^0x80) << 8; 3017 } 3018 } 3019 3020 frameCount = mFrameCount; 3021 int16_t *out = mMixBuffer; 3022 if (rampVolume) { 3023 if (mChannelCount == 1) { 3024 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3025 int32_t vlInc = d / (int32_t)frameCount; 3026 int32_t vl = ((int32_t)mLeftVolShort << 16); 3027 do { 3028 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3029 out++; 3030 vl += vlInc; 3031 } while (--frameCount); 3032 3033 } else { 3034 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3035 int32_t vlInc = d / (int32_t)frameCount; 3036 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3037 int32_t vrInc = d / (int32_t)frameCount; 3038 int32_t vl = ((int32_t)mLeftVolShort << 16); 3039 int32_t vr = ((int32_t)mRightVolShort << 16); 3040 do { 3041 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3042 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3043 out += 2; 3044 vl += vlInc; 3045 vr += vrInc; 3046 } while (--frameCount); 3047 } 3048 } else { 3049 if (mChannelCount == 1) { 3050 do { 3051 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3052 out++; 3053 } while (--frameCount); 3054 } else { 3055 do { 3056 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3057 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3058 out += 2; 3059 } while (--frameCount); 3060 } 3061 } 3062 3063 // convert back to unsigned 8 bit after volume calculation 3064 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3065 size_t count = mFrameCount * mChannelCount; 3066 int16_t *src = mMixBuffer; 3067 uint8_t *dst = (uint8_t *)mMixBuffer; 3068 while (count--) { 3069 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3070 } 3071 } 3072 3073 mLeftVolShort = leftVol; 3074 mRightVolShort = rightVol; 3075} 3076 3077void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3078{ 3079 if (sleepTime == 0) { 3080 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3081 sleepTime = activeSleepTime; 3082 } else { 3083 sleepTime = idleSleepTime; 3084 } 3085 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3086 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3087 sleepTime = 0; 3088 } 3089} 3090 3091// getTrackName_l() must be called with ThreadBase::mLock held 3092int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3093{ 3094 return 0; 3095} 3096 3097// deleteTrackName_l() must be called with ThreadBase::mLock held 3098void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3099{ 3100} 3101 3102// checkForNewParameters_l() must be called with ThreadBase::mLock held 3103bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3104{ 3105 bool reconfig = false; 3106 3107 while (!mNewParameters.isEmpty()) { 3108 status_t status = NO_ERROR; 3109 String8 keyValuePair = mNewParameters[0]; 3110 AudioParameter param = AudioParameter(keyValuePair); 3111 int value; 3112 3113 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3114 // do not accept frame count changes if tracks are open as the track buffer 3115 // size depends on frame count and correct behavior would not be garantied 3116 // if frame count is changed after track creation 3117 if (!mTracks.isEmpty()) { 3118 status = INVALID_OPERATION; 3119 } else { 3120 reconfig = true; 3121 } 3122 } 3123 if (status == NO_ERROR) { 3124 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3125 keyValuePair.string()); 3126 if (!mStandby && status == INVALID_OPERATION) { 3127 mOutput->stream->common.standby(&mOutput->stream->common); 3128 mStandby = true; 3129 mBytesWritten = 0; 3130 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3131 keyValuePair.string()); 3132 } 3133 if (status == NO_ERROR && reconfig) { 3134 readOutputParameters(); 3135 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3136 } 3137 } 3138 3139 mNewParameters.removeAt(0); 3140 3141 mParamStatus = status; 3142 mParamCond.signal(); 3143 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3144 // already timed out waiting for the status and will never signal the condition. 3145 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3146 } 3147 return reconfig; 3148} 3149 3150uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3151{ 3152 uint32_t time; 3153 if (audio_is_linear_pcm(mFormat)) { 3154 time = PlaybackThread::activeSleepTimeUs(); 3155 } else { 3156 time = 10000; 3157 } 3158 return time; 3159} 3160 3161uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3162{ 3163 uint32_t time; 3164 if (audio_is_linear_pcm(mFormat)) { 3165 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3166 } else { 3167 time = 10000; 3168 } 3169 return time; 3170} 3171 3172uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3173{ 3174 uint32_t time; 3175 if (audio_is_linear_pcm(mFormat)) { 3176 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3177 } else { 3178 time = 10000; 3179 } 3180 return time; 3181} 3182 3183void AudioFlinger::DirectOutputThread::cacheParameters_l() 3184{ 3185 PlaybackThread::cacheParameters_l(); 3186 3187 // use shorter standby delay as on normal output to release 3188 // hardware resources as soon as possible 3189 standbyDelay = microseconds(activeSleepTime*2); 3190} 3191 3192// ---------------------------------------------------------------------------- 3193 3194AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3195 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3196 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3197 mWaitTimeMs(UINT_MAX) 3198{ 3199 addOutputTrack(mainThread); 3200} 3201 3202AudioFlinger::DuplicatingThread::~DuplicatingThread() 3203{ 3204 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3205 mOutputTracks[i]->destroy(); 3206 } 3207} 3208 3209void AudioFlinger::DuplicatingThread::threadLoop_mix() 3210{ 3211 // mix buffers... 3212 if (outputsReady(outputTracks)) { 3213 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3214 } else { 3215 memset(mMixBuffer, 0, mixBufferSize); 3216 } 3217 sleepTime = 0; 3218 writeFrames = mFrameCount; 3219} 3220 3221void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3222{ 3223 if (sleepTime == 0) { 3224 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3225 sleepTime = activeSleepTime; 3226 } else { 3227 sleepTime = idleSleepTime; 3228 } 3229 } else if (mBytesWritten != 0) { 3230 // flush remaining overflow buffers in output tracks 3231 for (size_t i = 0; i < outputTracks.size(); i++) { 3232 if (outputTracks[i]->isActive()) { 3233 sleepTime = 0; 3234 writeFrames = 0; 3235 memset(mMixBuffer, 0, mixBufferSize); 3236 break; 3237 } 3238 } 3239 } 3240} 3241 3242void AudioFlinger::DuplicatingThread::threadLoop_write() 3243{ 3244 standbyTime = systemTime() + standbyDelay; 3245 for (size_t i = 0; i < outputTracks.size(); i++) { 3246 outputTracks[i]->write(mMixBuffer, writeFrames); 3247 } 3248 mBytesWritten += mixBufferSize; 3249} 3250 3251void AudioFlinger::DuplicatingThread::threadLoop_standby() 3252{ 3253 // DuplicatingThread implements standby by stopping all tracks 3254 for (size_t i = 0; i < outputTracks.size(); i++) { 3255 outputTracks[i]->stop(); 3256 } 3257} 3258 3259void AudioFlinger::DuplicatingThread::saveOutputTracks() 3260{ 3261 outputTracks = mOutputTracks; 3262} 3263 3264void AudioFlinger::DuplicatingThread::clearOutputTracks() 3265{ 3266 outputTracks.clear(); 3267} 3268 3269void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3270{ 3271 Mutex::Autolock _l(mLock); 3272 // FIXME explain this formula 3273 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3274 OutputTrack *outputTrack = new OutputTrack(thread, 3275 this, 3276 mSampleRate, 3277 mFormat, 3278 mChannelMask, 3279 frameCount); 3280 if (outputTrack->cblk() != NULL) { 3281 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3282 mOutputTracks.add(outputTrack); 3283 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3284 updateWaitTime_l(); 3285 } 3286} 3287 3288void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3289{ 3290 Mutex::Autolock _l(mLock); 3291 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3292 if (mOutputTracks[i]->thread() == thread) { 3293 mOutputTracks[i]->destroy(); 3294 mOutputTracks.removeAt(i); 3295 updateWaitTime_l(); 3296 return; 3297 } 3298 } 3299 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3300} 3301 3302// caller must hold mLock 3303void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3304{ 3305 mWaitTimeMs = UINT_MAX; 3306 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3307 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3308 if (strong != 0) { 3309 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3310 if (waitTimeMs < mWaitTimeMs) { 3311 mWaitTimeMs = waitTimeMs; 3312 } 3313 } 3314 } 3315} 3316 3317 3318bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3319{ 3320 for (size_t i = 0; i < outputTracks.size(); i++) { 3321 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3322 if (thread == 0) { 3323 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3324 return false; 3325 } 3326 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3327 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3328 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3329 return false; 3330 } 3331 } 3332 return true; 3333} 3334 3335uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3336{ 3337 return (mWaitTimeMs * 1000) / 2; 3338} 3339 3340void AudioFlinger::DuplicatingThread::cacheParameters_l() 3341{ 3342 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3343 updateWaitTime_l(); 3344 3345 MixerThread::cacheParameters_l(); 3346} 3347 3348// ---------------------------------------------------------------------------- 3349 3350// TrackBase constructor must be called with AudioFlinger::mLock held 3351AudioFlinger::ThreadBase::TrackBase::TrackBase( 3352 ThreadBase *thread, 3353 const sp<Client>& client, 3354 uint32_t sampleRate, 3355 audio_format_t format, 3356 uint32_t channelMask, 3357 int frameCount, 3358 const sp<IMemory>& sharedBuffer, 3359 int sessionId) 3360 : RefBase(), 3361 mThread(thread), 3362 mClient(client), 3363 mCblk(NULL), 3364 // mBuffer 3365 // mBufferEnd 3366 mFrameCount(0), 3367 mState(IDLE), 3368 mFormat(format), 3369 mStepServerFailed(false), 3370 mSessionId(sessionId) 3371 // mChannelCount 3372 // mChannelMask 3373{ 3374 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3375 3376 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3377 size_t size = sizeof(audio_track_cblk_t); 3378 uint8_t channelCount = popcount(channelMask); 3379 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3380 if (sharedBuffer == 0) { 3381 size += bufferSize; 3382 } 3383 3384 if (client != NULL) { 3385 mCblkMemory = client->heap()->allocate(size); 3386 if (mCblkMemory != 0) { 3387 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3388 if (mCblk != NULL) { // construct the shared structure in-place. 3389 new(mCblk) audio_track_cblk_t(); 3390 // clear all buffers 3391 mCblk->frameCount = frameCount; 3392 mCblk->sampleRate = sampleRate; 3393// uncomment the following lines to quickly test 32-bit wraparound 3394// mCblk->user = 0xffff0000; 3395// mCblk->server = 0xffff0000; 3396// mCblk->userBase = 0xffff0000; 3397// mCblk->serverBase = 0xffff0000; 3398 mChannelCount = channelCount; 3399 mChannelMask = channelMask; 3400 if (sharedBuffer == 0) { 3401 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3402 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3403 // Force underrun condition to avoid false underrun callback until first data is 3404 // written to buffer (other flags are cleared) 3405 mCblk->flags = CBLK_UNDERRUN_ON; 3406 } else { 3407 mBuffer = sharedBuffer->pointer(); 3408 } 3409 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3410 } 3411 } else { 3412 ALOGE("not enough memory for AudioTrack size=%u", size); 3413 client->heap()->dump("AudioTrack"); 3414 return; 3415 } 3416 } else { 3417 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3418 // construct the shared structure in-place. 3419 new(mCblk) audio_track_cblk_t(); 3420 // clear all buffers 3421 mCblk->frameCount = frameCount; 3422 mCblk->sampleRate = sampleRate; 3423// uncomment the following lines to quickly test 32-bit wraparound 3424// mCblk->user = 0xffff0000; 3425// mCblk->server = 0xffff0000; 3426// mCblk->userBase = 0xffff0000; 3427// mCblk->serverBase = 0xffff0000; 3428 mChannelCount = channelCount; 3429 mChannelMask = channelMask; 3430 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3431 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3432 // Force underrun condition to avoid false underrun callback until first data is 3433 // written to buffer (other flags are cleared) 3434 mCblk->flags = CBLK_UNDERRUN_ON; 3435 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3436 } 3437} 3438 3439AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3440{ 3441 if (mCblk != NULL) { 3442 if (mClient == 0) { 3443 delete mCblk; 3444 } else { 3445 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3446 } 3447 } 3448 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3449 if (mClient != 0) { 3450 // Client destructor must run with AudioFlinger mutex locked 3451 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3452 // If the client's reference count drops to zero, the associated destructor 3453 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3454 // relying on the automatic clear() at end of scope. 3455 mClient.clear(); 3456 } 3457} 3458 3459// AudioBufferProvider interface 3460// getNextBuffer() = 0; 3461// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3462void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3463{ 3464 buffer->raw = NULL; 3465 mFrameCount = buffer->frameCount; 3466 (void) step(); // ignore return value of step() 3467 buffer->frameCount = 0; 3468} 3469 3470bool AudioFlinger::ThreadBase::TrackBase::step() { 3471 bool result; 3472 audio_track_cblk_t* cblk = this->cblk(); 3473 3474 result = cblk->stepServer(mFrameCount); 3475 if (!result) { 3476 ALOGV("stepServer failed acquiring cblk mutex"); 3477 mStepServerFailed = true; 3478 } 3479 return result; 3480} 3481 3482void AudioFlinger::ThreadBase::TrackBase::reset() { 3483 audio_track_cblk_t* cblk = this->cblk(); 3484 3485 cblk->user = 0; 3486 cblk->server = 0; 3487 cblk->userBase = 0; 3488 cblk->serverBase = 0; 3489 mStepServerFailed = false; 3490 ALOGV("TrackBase::reset"); 3491} 3492 3493int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3494 return (int)mCblk->sampleRate; 3495} 3496 3497void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3498 audio_track_cblk_t* cblk = this->cblk(); 3499 size_t frameSize = cblk->frameSize; 3500 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3501 int8_t *bufferEnd = bufferStart + frames * frameSize; 3502 3503 // Check validity of returned pointer in case the track control block would have been corrupted. 3504 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 3505 "TrackBase::getBuffer buffer out of range:\n" 3506 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 3507 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 3508 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3509 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 3510 3511 return bufferStart; 3512} 3513 3514status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3515{ 3516 mSyncEvents.add(event); 3517 return NO_ERROR; 3518} 3519 3520// ---------------------------------------------------------------------------- 3521 3522// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3523AudioFlinger::PlaybackThread::Track::Track( 3524 PlaybackThread *thread, 3525 const sp<Client>& client, 3526 audio_stream_type_t streamType, 3527 uint32_t sampleRate, 3528 audio_format_t format, 3529 uint32_t channelMask, 3530 int frameCount, 3531 const sp<IMemory>& sharedBuffer, 3532 int sessionId, 3533 IAudioFlinger::track_flags_t flags) 3534 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3535 mMute(false), 3536 // mFillingUpStatus ? 3537 // mRetryCount initialized later when needed 3538 mSharedBuffer(sharedBuffer), 3539 mStreamType(streamType), 3540 mName(-1), // see note below 3541 mMainBuffer(thread->mixBuffer()), 3542 mAuxBuffer(NULL), 3543 mAuxEffectId(0), mHasVolumeController(false), 3544 mPresentationCompleteFrames(0), 3545 mFlags(flags) 3546{ 3547 if (mCblk != NULL) { 3548 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3549 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3550 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3551 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3552 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 3553 if (mName < 0) { 3554 ALOGE("no more track names available"); 3555 } 3556 } 3557 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3558} 3559 3560AudioFlinger::PlaybackThread::Track::~Track() 3561{ 3562 ALOGV("PlaybackThread::Track destructor"); 3563 sp<ThreadBase> thread = mThread.promote(); 3564 if (thread != 0) { 3565 Mutex::Autolock _l(thread->mLock); 3566 mState = TERMINATED; 3567 } 3568} 3569 3570void AudioFlinger::PlaybackThread::Track::destroy() 3571{ 3572 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3573 // by removing it from mTracks vector, so there is a risk that this Tracks's 3574 // destructor is called. As the destructor needs to lock mLock, 3575 // we must acquire a strong reference on this Track before locking mLock 3576 // here so that the destructor is called only when exiting this function. 3577 // On the other hand, as long as Track::destroy() is only called by 3578 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3579 // this Track with its member mTrack. 3580 sp<Track> keep(this); 3581 { // scope for mLock 3582 sp<ThreadBase> thread = mThread.promote(); 3583 if (thread != 0) { 3584 if (!isOutputTrack()) { 3585 if (mState == ACTIVE || mState == RESUMING) { 3586 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3587 3588#ifdef ADD_BATTERY_DATA 3589 // to track the speaker usage 3590 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3591#endif 3592 } 3593 AudioSystem::releaseOutput(thread->id()); 3594 } 3595 Mutex::Autolock _l(thread->mLock); 3596 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3597 playbackThread->destroyTrack_l(this); 3598 } 3599 } 3600} 3601 3602void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3603{ 3604 uint32_t vlr = mCblk->getVolumeLR(); 3605 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", 3606 mName - AudioMixer::TRACK0, 3607 (mClient == 0) ? getpid_cached : mClient->pid(), 3608 mStreamType, 3609 mFormat, 3610 mChannelMask, 3611 mSessionId, 3612 mFrameCount, 3613 mState, 3614 mMute, 3615 mFillingUpStatus, 3616 mCblk->sampleRate, 3617 vlr & 0xFFFF, 3618 vlr >> 16, 3619 mCblk->server, 3620 mCblk->user, 3621 (int)mMainBuffer, 3622 (int)mAuxBuffer); 3623} 3624 3625// AudioBufferProvider interface 3626status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3627 AudioBufferProvider::Buffer* buffer, int64_t pts) 3628{ 3629 audio_track_cblk_t* cblk = this->cblk(); 3630 uint32_t framesReady; 3631 uint32_t framesReq = buffer->frameCount; 3632 3633 // Check if last stepServer failed, try to step now 3634 if (mStepServerFailed) { 3635 if (!step()) goto getNextBuffer_exit; 3636 ALOGV("stepServer recovered"); 3637 mStepServerFailed = false; 3638 } 3639 3640 framesReady = cblk->framesReady(); 3641 3642 if (CC_LIKELY(framesReady)) { 3643 uint32_t s = cblk->server; 3644 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3645 3646 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3647 if (framesReq > framesReady) { 3648 framesReq = framesReady; 3649 } 3650 if (framesReq > bufferEnd - s) { 3651 framesReq = bufferEnd - s; 3652 } 3653 3654 buffer->raw = getBuffer(s, framesReq); 3655 if (buffer->raw == NULL) goto getNextBuffer_exit; 3656 3657 buffer->frameCount = framesReq; 3658 return NO_ERROR; 3659 } 3660 3661getNextBuffer_exit: 3662 buffer->raw = NULL; 3663 buffer->frameCount = 0; 3664 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3665 return NOT_ENOUGH_DATA; 3666} 3667 3668uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3669 return mCblk->framesReady(); 3670} 3671 3672bool AudioFlinger::PlaybackThread::Track::isReady() const { 3673 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3674 3675 if (framesReady() >= mCblk->frameCount || 3676 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3677 mFillingUpStatus = FS_FILLED; 3678 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3679 return true; 3680 } 3681 return false; 3682} 3683 3684status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid, 3685 AudioSystem::sync_event_t event, 3686 int triggerSession) 3687{ 3688 status_t status = NO_ERROR; 3689 ALOGV("start(%d), calling pid %d session %d tid %d", 3690 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3691 // check for use case 2 with missing callback 3692 if (isFastTrack() && (mSharedBuffer == 0) && (tid == 0)) { 3693 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied"); 3694 mFlags &= ~IAudioFlinger::TRACK_FAST; 3695 // FIXME the track must be invalidated and moved to another thread or 3696 // attached directly to the normal mixer now 3697 } 3698 sp<ThreadBase> thread = mThread.promote(); 3699 if (thread != 0) { 3700 Mutex::Autolock _l(thread->mLock); 3701 track_state state = mState; 3702 // here the track could be either new, or restarted 3703 // in both cases "unstop" the track 3704 if (mState == PAUSED) { 3705 mState = TrackBase::RESUMING; 3706 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3707 } else { 3708 mState = TrackBase::ACTIVE; 3709 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3710 } 3711 3712 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3713 thread->mLock.unlock(); 3714 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3715 thread->mLock.lock(); 3716 3717#ifdef ADD_BATTERY_DATA 3718 // to track the speaker usage 3719 if (status == NO_ERROR) { 3720 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3721 } 3722#endif 3723 } 3724 if (status == NO_ERROR) { 3725 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3726 playbackThread->addTrack_l(this); 3727 } else { 3728 mState = state; 3729 } 3730 } else { 3731 status = BAD_VALUE; 3732 } 3733 return status; 3734} 3735 3736void AudioFlinger::PlaybackThread::Track::stop() 3737{ 3738 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3739 sp<ThreadBase> thread = mThread.promote(); 3740 if (thread != 0) { 3741 Mutex::Autolock _l(thread->mLock); 3742 track_state state = mState; 3743 if (mState > STOPPED) { 3744 mState = STOPPED; 3745 // If the track is not active (PAUSED and buffers full), flush buffers 3746 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3747 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3748 reset(); 3749 } 3750 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3751 } 3752 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3753 thread->mLock.unlock(); 3754 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3755 thread->mLock.lock(); 3756 3757#ifdef ADD_BATTERY_DATA 3758 // to track the speaker usage 3759 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3760#endif 3761 } 3762 } 3763} 3764 3765void AudioFlinger::PlaybackThread::Track::pause() 3766{ 3767 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3768 sp<ThreadBase> thread = mThread.promote(); 3769 if (thread != 0) { 3770 Mutex::Autolock _l(thread->mLock); 3771 if (mState == ACTIVE || mState == RESUMING) { 3772 mState = PAUSING; 3773 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3774 if (!isOutputTrack()) { 3775 thread->mLock.unlock(); 3776 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3777 thread->mLock.lock(); 3778 3779#ifdef ADD_BATTERY_DATA 3780 // to track the speaker usage 3781 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3782#endif 3783 } 3784 } 3785 } 3786} 3787 3788void AudioFlinger::PlaybackThread::Track::flush() 3789{ 3790 ALOGV("flush(%d)", mName); 3791 sp<ThreadBase> thread = mThread.promote(); 3792 if (thread != 0) { 3793 Mutex::Autolock _l(thread->mLock); 3794 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3795 return; 3796 } 3797 // No point remaining in PAUSED state after a flush => go to 3798 // STOPPED state 3799 mState = STOPPED; 3800 3801 // do not reset the track if it is still in the process of being stopped or paused. 3802 // this will be done by prepareTracks_l() when the track is stopped. 3803 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3804 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3805 reset(); 3806 } 3807 } 3808} 3809 3810void AudioFlinger::PlaybackThread::Track::reset() 3811{ 3812 // Do not reset twice to avoid discarding data written just after a flush and before 3813 // the audioflinger thread detects the track is stopped. 3814 if (!mResetDone) { 3815 TrackBase::reset(); 3816 // Force underrun condition to avoid false underrun callback until first data is 3817 // written to buffer 3818 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3819 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3820 mFillingUpStatus = FS_FILLING; 3821 mResetDone = true; 3822 mPresentationCompleteFrames = 0; 3823 } 3824} 3825 3826void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3827{ 3828 mMute = muted; 3829} 3830 3831status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3832{ 3833 status_t status = DEAD_OBJECT; 3834 sp<ThreadBase> thread = mThread.promote(); 3835 if (thread != 0) { 3836 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3837 status = playbackThread->attachAuxEffect(this, EffectId); 3838 } 3839 return status; 3840} 3841 3842void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3843{ 3844 mAuxEffectId = EffectId; 3845 mAuxBuffer = buffer; 3846} 3847 3848bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 3849 size_t audioHalFrames) 3850{ 3851 // a track is considered presented when the total number of frames written to audio HAL 3852 // corresponds to the number of frames written when presentationComplete() is called for the 3853 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 3854 if (mPresentationCompleteFrames == 0) { 3855 mPresentationCompleteFrames = framesWritten + audioHalFrames; 3856 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 3857 mPresentationCompleteFrames, audioHalFrames); 3858 } 3859 if (framesWritten >= mPresentationCompleteFrames) { 3860 ALOGV("presentationComplete() session %d complete: framesWritten %d", 3861 mSessionId, framesWritten); 3862 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 3863 mPresentationCompleteFrames = 0; 3864 return true; 3865 } 3866 return false; 3867} 3868 3869void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 3870{ 3871 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 3872 if (mSyncEvents[i]->type() == type) { 3873 mSyncEvents[i]->trigger(); 3874 mSyncEvents.removeAt(i); 3875 i--; 3876 } 3877 } 3878} 3879 3880 3881// timed audio tracks 3882 3883sp<AudioFlinger::PlaybackThread::TimedTrack> 3884AudioFlinger::PlaybackThread::TimedTrack::create( 3885 PlaybackThread *thread, 3886 const sp<Client>& client, 3887 audio_stream_type_t streamType, 3888 uint32_t sampleRate, 3889 audio_format_t format, 3890 uint32_t channelMask, 3891 int frameCount, 3892 const sp<IMemory>& sharedBuffer, 3893 int sessionId) { 3894 if (!client->reserveTimedTrack()) 3895 return NULL; 3896 3897 return new TimedTrack( 3898 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3899 sharedBuffer, sessionId); 3900} 3901 3902AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3903 PlaybackThread *thread, 3904 const sp<Client>& client, 3905 audio_stream_type_t streamType, 3906 uint32_t sampleRate, 3907 audio_format_t format, 3908 uint32_t channelMask, 3909 int frameCount, 3910 const sp<IMemory>& sharedBuffer, 3911 int sessionId) 3912 : Track(thread, client, streamType, sampleRate, format, channelMask, 3913 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 3914 mQueueHeadInFlight(false), 3915 mTrimQueueHeadOnRelease(false), 3916 mFramesPendingInQueue(0), 3917 mTimedSilenceBuffer(NULL), 3918 mTimedSilenceBufferSize(0), 3919 mTimedAudioOutputOnTime(false), 3920 mMediaTimeTransformValid(false) 3921{ 3922 LocalClock lc; 3923 mLocalTimeFreq = lc.getLocalFreq(); 3924 3925 mLocalTimeToSampleTransform.a_zero = 0; 3926 mLocalTimeToSampleTransform.b_zero = 0; 3927 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3928 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3929 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3930 &mLocalTimeToSampleTransform.a_to_b_denom); 3931 3932 mMediaTimeToSampleTransform.a_zero = 0; 3933 mMediaTimeToSampleTransform.b_zero = 0; 3934 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 3935 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 3936 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 3937 &mMediaTimeToSampleTransform.a_to_b_denom); 3938} 3939 3940AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3941 mClient->releaseTimedTrack(); 3942 delete [] mTimedSilenceBuffer; 3943} 3944 3945status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3946 size_t size, sp<IMemory>* buffer) { 3947 3948 Mutex::Autolock _l(mTimedBufferQueueLock); 3949 3950 trimTimedBufferQueue_l(); 3951 3952 // lazily initialize the shared memory heap for timed buffers 3953 if (mTimedMemoryDealer == NULL) { 3954 const int kTimedBufferHeapSize = 512 << 10; 3955 3956 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3957 "AudioFlingerTimed"); 3958 if (mTimedMemoryDealer == NULL) 3959 return NO_MEMORY; 3960 } 3961 3962 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3963 if (newBuffer == NULL) { 3964 newBuffer = mTimedMemoryDealer->allocate(size); 3965 if (newBuffer == NULL) 3966 return NO_MEMORY; 3967 } 3968 3969 *buffer = newBuffer; 3970 return NO_ERROR; 3971} 3972 3973// caller must hold mTimedBufferQueueLock 3974void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3975 int64_t mediaTimeNow; 3976 { 3977 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3978 if (!mMediaTimeTransformValid) 3979 return; 3980 3981 int64_t targetTimeNow; 3982 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3983 ? mCCHelper.getCommonTime(&targetTimeNow) 3984 : mCCHelper.getLocalTime(&targetTimeNow); 3985 3986 if (OK != res) 3987 return; 3988 3989 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3990 &mediaTimeNow)) { 3991 return; 3992 } 3993 } 3994 3995 size_t trimEnd; 3996 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 3997 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 3998 / mCblk->frameSize; 3999 int64_t bufEnd; 4000 4001 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4002 &bufEnd)) { 4003 ALOGE("Failed to convert frame count of %lld to media time duration" 4004 " (scale factor %d/%u) in %s", frameCount, 4005 mMediaTimeToSampleTransform.a_to_b_numer, 4006 mMediaTimeToSampleTransform.a_to_b_denom, 4007 __PRETTY_FUNCTION__); 4008 break; 4009 } 4010 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4011 4012 if (bufEnd > mediaTimeNow) 4013 break; 4014 4015 // Is the buffer we want to use in the middle of a mix operation right 4016 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4017 // from the mixer which should be coming back shortly. 4018 if (!trimEnd && mQueueHeadInFlight) { 4019 mTrimQueueHeadOnRelease = true; 4020 } 4021 } 4022 4023 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4024 if (trimStart < trimEnd) { 4025 // Update the bookkeeping for framesReady() 4026 for (size_t i = trimStart; i < trimEnd; ++i) { 4027 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4028 } 4029 4030 // Now actually remove the buffers from the queue. 4031 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4032 } 4033} 4034 4035void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4036 const char* logTag) { 4037 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4038 "%s called (reason \"%s\"), but timed buffer queue has no" 4039 " elements to trim.", __FUNCTION__, logTag); 4040 4041 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4042 mTimedBufferQueue.removeAt(0); 4043} 4044 4045void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4046 const TimedBuffer& buf, 4047 const char* logTag) { 4048 uint32_t bufBytes = buf.buffer()->size(); 4049 uint32_t consumedAlready = buf.position(); 4050 4051 ALOG_ASSERT(consumedAlready <= bufBytes, 4052 "Bad bookkeeping while updating frames pending. Timed buffer is" 4053 " only %u bytes long, but claims to have consumed %u" 4054 " bytes. (update reason: \"%s\")", 4055 bufBytes, consumedAlready, logTag); 4056 4057 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4058 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4059 "Bad bookkeeping while updating frames pending. Should have at" 4060 " least %u queued frames, but we think we have only %u. (update" 4061 " reason: \"%s\")", 4062 bufFrames, mFramesPendingInQueue, logTag); 4063 4064 mFramesPendingInQueue -= bufFrames; 4065} 4066 4067status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4068 const sp<IMemory>& buffer, int64_t pts) { 4069 4070 { 4071 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4072 if (!mMediaTimeTransformValid) 4073 return INVALID_OPERATION; 4074 } 4075 4076 Mutex::Autolock _l(mTimedBufferQueueLock); 4077 4078 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4079 mFramesPendingInQueue += bufFrames; 4080 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4081 4082 return NO_ERROR; 4083} 4084 4085status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4086 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4087 4088 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4089 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4090 target); 4091 4092 if (!(target == TimedAudioTrack::LOCAL_TIME || 4093 target == TimedAudioTrack::COMMON_TIME)) { 4094 return BAD_VALUE; 4095 } 4096 4097 Mutex::Autolock lock(mMediaTimeTransformLock); 4098 mMediaTimeTransform = xform; 4099 mMediaTimeTransformTarget = target; 4100 mMediaTimeTransformValid = true; 4101 4102 return NO_ERROR; 4103} 4104 4105#define min(a, b) ((a) < (b) ? (a) : (b)) 4106 4107// implementation of getNextBuffer for tracks whose buffers have timestamps 4108status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4109 AudioBufferProvider::Buffer* buffer, int64_t pts) 4110{ 4111 if (pts == AudioBufferProvider::kInvalidPTS) { 4112 buffer->raw = 0; 4113 buffer->frameCount = 0; 4114 return INVALID_OPERATION; 4115 } 4116 4117 Mutex::Autolock _l(mTimedBufferQueueLock); 4118 4119 ALOG_ASSERT(!mQueueHeadInFlight, 4120 "getNextBuffer called without releaseBuffer!"); 4121 4122 while (true) { 4123 4124 // if we have no timed buffers, then fail 4125 if (mTimedBufferQueue.isEmpty()) { 4126 buffer->raw = 0; 4127 buffer->frameCount = 0; 4128 return NOT_ENOUGH_DATA; 4129 } 4130 4131 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4132 4133 // calculate the PTS of the head of the timed buffer queue expressed in 4134 // local time 4135 int64_t headLocalPTS; 4136 { 4137 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4138 4139 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4140 4141 if (mMediaTimeTransform.a_to_b_denom == 0) { 4142 // the transform represents a pause, so yield silence 4143 timedYieldSilence_l(buffer->frameCount, buffer); 4144 return NO_ERROR; 4145 } 4146 4147 int64_t transformedPTS; 4148 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4149 &transformedPTS)) { 4150 // the transform failed. this shouldn't happen, but if it does 4151 // then just drop this buffer 4152 ALOGW("timedGetNextBuffer transform failed"); 4153 buffer->raw = 0; 4154 buffer->frameCount = 0; 4155 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4156 return NO_ERROR; 4157 } 4158 4159 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4160 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4161 &headLocalPTS)) { 4162 buffer->raw = 0; 4163 buffer->frameCount = 0; 4164 return INVALID_OPERATION; 4165 } 4166 } else { 4167 headLocalPTS = transformedPTS; 4168 } 4169 } 4170 4171 // adjust the head buffer's PTS to reflect the portion of the head buffer 4172 // that has already been consumed 4173 int64_t effectivePTS = headLocalPTS + 4174 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4175 4176 // Calculate the delta in samples between the head of the input buffer 4177 // queue and the start of the next output buffer that will be written. 4178 // If the transformation fails because of over or underflow, it means 4179 // that the sample's position in the output stream is so far out of 4180 // whack that it should just be dropped. 4181 int64_t sampleDelta; 4182 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4183 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4184 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4185 " mix"); 4186 continue; 4187 } 4188 if (!mLocalTimeToSampleTransform.doForwardTransform( 4189 (effectivePTS - pts) << 32, &sampleDelta)) { 4190 ALOGV("*** too late during sample rate transform: dropped buffer"); 4191 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 4192 continue; 4193 } 4194 4195 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 4196 " sampleDelta=[%d.%08x]", 4197 head.pts(), head.position(), pts, 4198 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 4199 + (sampleDelta >> 32)), 4200 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4201 4202 // if the delta between the ideal placement for the next input sample and 4203 // the current output position is within this threshold, then we will 4204 // concatenate the next input samples to the previous output 4205 const int64_t kSampleContinuityThreshold = 4206 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4207 4208 // if this is the first buffer of audio that we're emitting from this track 4209 // then it should be almost exactly on time. 4210 const int64_t kSampleStartupThreshold = 1LL << 32; 4211 4212 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4213 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4214 // the next input is close enough to being on time, so concatenate it 4215 // with the last output 4216 timedYieldSamples_l(buffer); 4217 4218 ALOGVV("*** on time: head.pos=%d frameCount=%u", 4219 head.position(), buffer->frameCount); 4220 return NO_ERROR; 4221 } else if (sampleDelta > 0) { 4222 // the gap between the current output position and the proper start of 4223 // the next input sample is too big, so fill it with silence 4224 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4225 4226 timedYieldSilence_l(framesUntilNextInput, buffer); 4227 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4228 return NO_ERROR; 4229 } else { 4230 // the next input sample is late 4231 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4232 size_t onTimeSamplePosition = 4233 head.position() + lateFrames * mCblk->frameSize; 4234 4235 if (onTimeSamplePosition > head.buffer()->size()) { 4236 // all the remaining samples in the head are too late, so 4237 // drop it and move on 4238 ALOGV("*** too late: dropped buffer"); 4239 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 4240 continue; 4241 } else { 4242 // skip over the late samples 4243 head.setPosition(onTimeSamplePosition); 4244 4245 // yield the available samples 4246 timedYieldSamples_l(buffer); 4247 4248 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4249 return NO_ERROR; 4250 } 4251 } 4252 } 4253} 4254 4255// Yield samples from the timed buffer queue head up to the given output 4256// buffer's capacity. 4257// 4258// Caller must hold mTimedBufferQueueLock 4259void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 4260 AudioBufferProvider::Buffer* buffer) { 4261 4262 const TimedBuffer& head = mTimedBufferQueue[0]; 4263 4264 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4265 head.position()); 4266 4267 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4268 mCblk->frameSize); 4269 size_t framesRequested = buffer->frameCount; 4270 buffer->frameCount = min(framesLeftInHead, framesRequested); 4271 4272 mQueueHeadInFlight = true; 4273 mTimedAudioOutputOnTime = true; 4274} 4275 4276// Yield samples of silence up to the given output buffer's capacity 4277// 4278// Caller must hold mTimedBufferQueueLock 4279void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 4280 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4281 4282 // lazily allocate a buffer filled with silence 4283 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4284 delete [] mTimedSilenceBuffer; 4285 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4286 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4287 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4288 } 4289 4290 buffer->raw = mTimedSilenceBuffer; 4291 size_t framesRequested = buffer->frameCount; 4292 buffer->frameCount = min(numFrames, framesRequested); 4293 4294 mTimedAudioOutputOnTime = false; 4295} 4296 4297// AudioBufferProvider interface 4298void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4299 AudioBufferProvider::Buffer* buffer) { 4300 4301 Mutex::Autolock _l(mTimedBufferQueueLock); 4302 4303 // If the buffer which was just released is part of the buffer at the head 4304 // of the queue, be sure to update the amt of the buffer which has been 4305 // consumed. If the buffer being returned is not part of the head of the 4306 // queue, its either because the buffer is part of the silence buffer, or 4307 // because the head of the timed queue was trimmed after the mixer called 4308 // getNextBuffer but before the mixer called releaseBuffer. 4309 if (buffer->raw == mTimedSilenceBuffer) { 4310 ALOG_ASSERT(!mQueueHeadInFlight, 4311 "Queue head in flight during release of silence buffer!"); 4312 goto done; 4313 } 4314 4315 ALOG_ASSERT(mQueueHeadInFlight, 4316 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 4317 " head in flight."); 4318 4319 if (mTimedBufferQueue.size()) { 4320 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4321 4322 void* start = head.buffer()->pointer(); 4323 void* end = reinterpret_cast<void*>( 4324 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 4325 + head.buffer()->size()); 4326 4327 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 4328 "released buffer not within the head of the timed buffer" 4329 " queue; qHead = [%p, %p], released buffer = %p", 4330 start, end, buffer->raw); 4331 4332 head.setPosition(head.position() + 4333 (buffer->frameCount * mCblk->frameSize)); 4334 mQueueHeadInFlight = false; 4335 4336 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 4337 "Bad bookkeeping during releaseBuffer! Should have at" 4338 " least %u queued frames, but we think we have only %u", 4339 buffer->frameCount, mFramesPendingInQueue); 4340 4341 mFramesPendingInQueue -= buffer->frameCount; 4342 4343 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 4344 || mTrimQueueHeadOnRelease) { 4345 trimTimedBufferQueueHead_l("releaseBuffer"); 4346 mTrimQueueHeadOnRelease = false; 4347 } 4348 } else { 4349 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 4350 " buffers in the timed buffer queue"); 4351 } 4352 4353done: 4354 buffer->raw = 0; 4355 buffer->frameCount = 0; 4356} 4357 4358uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4359 Mutex::Autolock _l(mTimedBufferQueueLock); 4360 return mFramesPendingInQueue; 4361} 4362 4363AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4364 : mPTS(0), mPosition(0) {} 4365 4366AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4367 const sp<IMemory>& buffer, int64_t pts) 4368 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4369 4370// ---------------------------------------------------------------------------- 4371 4372// RecordTrack constructor must be called with AudioFlinger::mLock held 4373AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4374 RecordThread *thread, 4375 const sp<Client>& client, 4376 uint32_t sampleRate, 4377 audio_format_t format, 4378 uint32_t channelMask, 4379 int frameCount, 4380 int sessionId) 4381 : TrackBase(thread, client, sampleRate, format, 4382 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4383 mOverflow(false) 4384{ 4385 if (mCblk != NULL) { 4386 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4387 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4388 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4389 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4390 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4391 } else { 4392 mCblk->frameSize = sizeof(int8_t); 4393 } 4394 } 4395} 4396 4397AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4398{ 4399 sp<ThreadBase> thread = mThread.promote(); 4400 if (thread != 0) { 4401 AudioSystem::releaseInput(thread->id()); 4402 } 4403} 4404 4405// AudioBufferProvider interface 4406status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4407{ 4408 audio_track_cblk_t* cblk = this->cblk(); 4409 uint32_t framesAvail; 4410 uint32_t framesReq = buffer->frameCount; 4411 4412 // Check if last stepServer failed, try to step now 4413 if (mStepServerFailed) { 4414 if (!step()) goto getNextBuffer_exit; 4415 ALOGV("stepServer recovered"); 4416 mStepServerFailed = false; 4417 } 4418 4419 framesAvail = cblk->framesAvailable_l(); 4420 4421 if (CC_LIKELY(framesAvail)) { 4422 uint32_t s = cblk->server; 4423 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4424 4425 if (framesReq > framesAvail) { 4426 framesReq = framesAvail; 4427 } 4428 if (framesReq > bufferEnd - s) { 4429 framesReq = bufferEnd - s; 4430 } 4431 4432 buffer->raw = getBuffer(s, framesReq); 4433 if (buffer->raw == NULL) goto getNextBuffer_exit; 4434 4435 buffer->frameCount = framesReq; 4436 return NO_ERROR; 4437 } 4438 4439getNextBuffer_exit: 4440 buffer->raw = NULL; 4441 buffer->frameCount = 0; 4442 return NOT_ENOUGH_DATA; 4443} 4444 4445status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid, 4446 AudioSystem::sync_event_t event, 4447 int triggerSession) 4448{ 4449 sp<ThreadBase> thread = mThread.promote(); 4450 if (thread != 0) { 4451 RecordThread *recordThread = (RecordThread *)thread.get(); 4452 return recordThread->start(this, tid, event, triggerSession); 4453 } else { 4454 return BAD_VALUE; 4455 } 4456} 4457 4458void AudioFlinger::RecordThread::RecordTrack::stop() 4459{ 4460 sp<ThreadBase> thread = mThread.promote(); 4461 if (thread != 0) { 4462 RecordThread *recordThread = (RecordThread *)thread.get(); 4463 recordThread->stop(this); 4464 TrackBase::reset(); 4465 // Force overrun condition to avoid false overrun callback until first data is 4466 // read from buffer 4467 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4468 } 4469} 4470 4471void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4472{ 4473 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4474 (mClient == 0) ? getpid_cached : mClient->pid(), 4475 mFormat, 4476 mChannelMask, 4477 mSessionId, 4478 mFrameCount, 4479 mState, 4480 mCblk->sampleRate, 4481 mCblk->server, 4482 mCblk->user); 4483} 4484 4485 4486// ---------------------------------------------------------------------------- 4487 4488AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4489 PlaybackThread *playbackThread, 4490 DuplicatingThread *sourceThread, 4491 uint32_t sampleRate, 4492 audio_format_t format, 4493 uint32_t channelMask, 4494 int frameCount) 4495 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 4496 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 4497 mActive(false), mSourceThread(sourceThread) 4498{ 4499 4500 if (mCblk != NULL) { 4501 mCblk->flags |= CBLK_DIRECTION_OUT; 4502 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4503 mOutBuffer.frameCount = 0; 4504 playbackThread->mTracks.add(this); 4505 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4506 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4507 mCblk, mBuffer, mCblk->buffers, 4508 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4509 } else { 4510 ALOGW("Error creating output track on thread %p", playbackThread); 4511 } 4512} 4513 4514AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4515{ 4516 clearBufferQueue(); 4517} 4518 4519status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid, 4520 AudioSystem::sync_event_t event, 4521 int triggerSession) 4522{ 4523 status_t status = Track::start(tid, event, triggerSession); 4524 if (status != NO_ERROR) { 4525 return status; 4526 } 4527 4528 mActive = true; 4529 mRetryCount = 127; 4530 return status; 4531} 4532 4533void AudioFlinger::PlaybackThread::OutputTrack::stop() 4534{ 4535 Track::stop(); 4536 clearBufferQueue(); 4537 mOutBuffer.frameCount = 0; 4538 mActive = false; 4539} 4540 4541bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4542{ 4543 Buffer *pInBuffer; 4544 Buffer inBuffer; 4545 uint32_t channelCount = mChannelCount; 4546 bool outputBufferFull = false; 4547 inBuffer.frameCount = frames; 4548 inBuffer.i16 = data; 4549 4550 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4551 4552 if (!mActive && frames != 0) { 4553 start(0); 4554 sp<ThreadBase> thread = mThread.promote(); 4555 if (thread != 0) { 4556 MixerThread *mixerThread = (MixerThread *)thread.get(); 4557 if (mCblk->frameCount > frames){ 4558 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4559 uint32_t startFrames = (mCblk->frameCount - frames); 4560 pInBuffer = new Buffer; 4561 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4562 pInBuffer->frameCount = startFrames; 4563 pInBuffer->i16 = pInBuffer->mBuffer; 4564 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4565 mBufferQueue.add(pInBuffer); 4566 } else { 4567 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4568 } 4569 } 4570 } 4571 } 4572 4573 while (waitTimeLeftMs) { 4574 // First write pending buffers, then new data 4575 if (mBufferQueue.size()) { 4576 pInBuffer = mBufferQueue.itemAt(0); 4577 } else { 4578 pInBuffer = &inBuffer; 4579 } 4580 4581 if (pInBuffer->frameCount == 0) { 4582 break; 4583 } 4584 4585 if (mOutBuffer.frameCount == 0) { 4586 mOutBuffer.frameCount = pInBuffer->frameCount; 4587 nsecs_t startTime = systemTime(); 4588 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4589 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4590 outputBufferFull = true; 4591 break; 4592 } 4593 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4594 if (waitTimeLeftMs >= waitTimeMs) { 4595 waitTimeLeftMs -= waitTimeMs; 4596 } else { 4597 waitTimeLeftMs = 0; 4598 } 4599 } 4600 4601 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4602 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4603 mCblk->stepUser(outFrames); 4604 pInBuffer->frameCount -= outFrames; 4605 pInBuffer->i16 += outFrames * channelCount; 4606 mOutBuffer.frameCount -= outFrames; 4607 mOutBuffer.i16 += outFrames * channelCount; 4608 4609 if (pInBuffer->frameCount == 0) { 4610 if (mBufferQueue.size()) { 4611 mBufferQueue.removeAt(0); 4612 delete [] pInBuffer->mBuffer; 4613 delete pInBuffer; 4614 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4615 } else { 4616 break; 4617 } 4618 } 4619 } 4620 4621 // If we could not write all frames, allocate a buffer and queue it for next time. 4622 if (inBuffer.frameCount) { 4623 sp<ThreadBase> thread = mThread.promote(); 4624 if (thread != 0 && !thread->standby()) { 4625 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4626 pInBuffer = new Buffer; 4627 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4628 pInBuffer->frameCount = inBuffer.frameCount; 4629 pInBuffer->i16 = pInBuffer->mBuffer; 4630 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4631 mBufferQueue.add(pInBuffer); 4632 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4633 } else { 4634 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4635 } 4636 } 4637 } 4638 4639 // Calling write() with a 0 length buffer, means that no more data will be written: 4640 // If no more buffers are pending, fill output track buffer to make sure it is started 4641 // by output mixer. 4642 if (frames == 0 && mBufferQueue.size() == 0) { 4643 if (mCblk->user < mCblk->frameCount) { 4644 frames = mCblk->frameCount - mCblk->user; 4645 pInBuffer = new Buffer; 4646 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4647 pInBuffer->frameCount = frames; 4648 pInBuffer->i16 = pInBuffer->mBuffer; 4649 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4650 mBufferQueue.add(pInBuffer); 4651 } else if (mActive) { 4652 stop(); 4653 } 4654 } 4655 4656 return outputBufferFull; 4657} 4658 4659status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4660{ 4661 int active; 4662 status_t result; 4663 audio_track_cblk_t* cblk = mCblk; 4664 uint32_t framesReq = buffer->frameCount; 4665 4666// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4667 buffer->frameCount = 0; 4668 4669 uint32_t framesAvail = cblk->framesAvailable(); 4670 4671 4672 if (framesAvail == 0) { 4673 Mutex::Autolock _l(cblk->lock); 4674 goto start_loop_here; 4675 while (framesAvail == 0) { 4676 active = mActive; 4677 if (CC_UNLIKELY(!active)) { 4678 ALOGV("Not active and NO_MORE_BUFFERS"); 4679 return NO_MORE_BUFFERS; 4680 } 4681 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4682 if (result != NO_ERROR) { 4683 return NO_MORE_BUFFERS; 4684 } 4685 // read the server count again 4686 start_loop_here: 4687 framesAvail = cblk->framesAvailable_l(); 4688 } 4689 } 4690 4691// if (framesAvail < framesReq) { 4692// return NO_MORE_BUFFERS; 4693// } 4694 4695 if (framesReq > framesAvail) { 4696 framesReq = framesAvail; 4697 } 4698 4699 uint32_t u = cblk->user; 4700 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4701 4702 if (framesReq > bufferEnd - u) { 4703 framesReq = bufferEnd - u; 4704 } 4705 4706 buffer->frameCount = framesReq; 4707 buffer->raw = (void *)cblk->buffer(u); 4708 return NO_ERROR; 4709} 4710 4711 4712void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4713{ 4714 size_t size = mBufferQueue.size(); 4715 4716 for (size_t i = 0; i < size; i++) { 4717 Buffer *pBuffer = mBufferQueue.itemAt(i); 4718 delete [] pBuffer->mBuffer; 4719 delete pBuffer; 4720 } 4721 mBufferQueue.clear(); 4722} 4723 4724// ---------------------------------------------------------------------------- 4725 4726AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4727 : RefBase(), 4728 mAudioFlinger(audioFlinger), 4729 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4730 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4731 mPid(pid), 4732 mTimedTrackCount(0) 4733{ 4734 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4735} 4736 4737// Client destructor must be called with AudioFlinger::mLock held 4738AudioFlinger::Client::~Client() 4739{ 4740 mAudioFlinger->removeClient_l(mPid); 4741} 4742 4743sp<MemoryDealer> AudioFlinger::Client::heap() const 4744{ 4745 return mMemoryDealer; 4746} 4747 4748// Reserve one of the limited slots for a timed audio track associated 4749// with this client 4750bool AudioFlinger::Client::reserveTimedTrack() 4751{ 4752 const int kMaxTimedTracksPerClient = 4; 4753 4754 Mutex::Autolock _l(mTimedTrackLock); 4755 4756 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4757 ALOGW("can not create timed track - pid %d has exceeded the limit", 4758 mPid); 4759 return false; 4760 } 4761 4762 mTimedTrackCount++; 4763 return true; 4764} 4765 4766// Release a slot for a timed audio track 4767void AudioFlinger::Client::releaseTimedTrack() 4768{ 4769 Mutex::Autolock _l(mTimedTrackLock); 4770 mTimedTrackCount--; 4771} 4772 4773// ---------------------------------------------------------------------------- 4774 4775AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4776 const sp<IAudioFlingerClient>& client, 4777 pid_t pid) 4778 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4779{ 4780} 4781 4782AudioFlinger::NotificationClient::~NotificationClient() 4783{ 4784} 4785 4786void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4787{ 4788 sp<NotificationClient> keep(this); 4789 mAudioFlinger->removeNotificationClient(mPid); 4790} 4791 4792// ---------------------------------------------------------------------------- 4793 4794AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4795 : BnAudioTrack(), 4796 mTrack(track) 4797{ 4798} 4799 4800AudioFlinger::TrackHandle::~TrackHandle() { 4801 // just stop the track on deletion, associated resources 4802 // will be freed from the main thread once all pending buffers have 4803 // been played. Unless it's not in the active track list, in which 4804 // case we free everything now... 4805 mTrack->destroy(); 4806} 4807 4808sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4809 return mTrack->getCblk(); 4810} 4811 4812status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4813 return mTrack->start(tid); 4814} 4815 4816void AudioFlinger::TrackHandle::stop() { 4817 mTrack->stop(); 4818} 4819 4820void AudioFlinger::TrackHandle::flush() { 4821 mTrack->flush(); 4822} 4823 4824void AudioFlinger::TrackHandle::mute(bool e) { 4825 mTrack->mute(e); 4826} 4827 4828void AudioFlinger::TrackHandle::pause() { 4829 mTrack->pause(); 4830} 4831 4832status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4833{ 4834 return mTrack->attachAuxEffect(EffectId); 4835} 4836 4837status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4838 sp<IMemory>* buffer) { 4839 if (!mTrack->isTimedTrack()) 4840 return INVALID_OPERATION; 4841 4842 PlaybackThread::TimedTrack* tt = 4843 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4844 return tt->allocateTimedBuffer(size, buffer); 4845} 4846 4847status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4848 int64_t pts) { 4849 if (!mTrack->isTimedTrack()) 4850 return INVALID_OPERATION; 4851 4852 PlaybackThread::TimedTrack* tt = 4853 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4854 return tt->queueTimedBuffer(buffer, pts); 4855} 4856 4857status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4858 const LinearTransform& xform, int target) { 4859 4860 if (!mTrack->isTimedTrack()) 4861 return INVALID_OPERATION; 4862 4863 PlaybackThread::TimedTrack* tt = 4864 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4865 return tt->setMediaTimeTransform( 4866 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4867} 4868 4869status_t AudioFlinger::TrackHandle::onTransact( 4870 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4871{ 4872 return BnAudioTrack::onTransact(code, data, reply, flags); 4873} 4874 4875// ---------------------------------------------------------------------------- 4876 4877sp<IAudioRecord> AudioFlinger::openRecord( 4878 pid_t pid, 4879 audio_io_handle_t input, 4880 uint32_t sampleRate, 4881 audio_format_t format, 4882 uint32_t channelMask, 4883 int frameCount, 4884 IAudioFlinger::track_flags_t flags, 4885 int *sessionId, 4886 status_t *status) 4887{ 4888 sp<RecordThread::RecordTrack> recordTrack; 4889 sp<RecordHandle> recordHandle; 4890 sp<Client> client; 4891 status_t lStatus; 4892 RecordThread *thread; 4893 size_t inFrameCount; 4894 int lSessionId; 4895 4896 // check calling permissions 4897 if (!recordingAllowed()) { 4898 lStatus = PERMISSION_DENIED; 4899 goto Exit; 4900 } 4901 4902 // add client to list 4903 { // scope for mLock 4904 Mutex::Autolock _l(mLock); 4905 thread = checkRecordThread_l(input); 4906 if (thread == NULL) { 4907 lStatus = BAD_VALUE; 4908 goto Exit; 4909 } 4910 4911 client = registerPid_l(pid); 4912 4913 // If no audio session id is provided, create one here 4914 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4915 lSessionId = *sessionId; 4916 } else { 4917 lSessionId = nextUniqueId(); 4918 if (sessionId != NULL) { 4919 *sessionId = lSessionId; 4920 } 4921 } 4922 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4923 recordTrack = thread->createRecordTrack_l(client, 4924 sampleRate, 4925 format, 4926 channelMask, 4927 frameCount, 4928 lSessionId, 4929 &lStatus); 4930 } 4931 if (lStatus != NO_ERROR) { 4932 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4933 // destructor is called by the TrackBase destructor with mLock held 4934 client.clear(); 4935 recordTrack.clear(); 4936 goto Exit; 4937 } 4938 4939 // return to handle to client 4940 recordHandle = new RecordHandle(recordTrack); 4941 lStatus = NO_ERROR; 4942 4943Exit: 4944 if (status) { 4945 *status = lStatus; 4946 } 4947 return recordHandle; 4948} 4949 4950// ---------------------------------------------------------------------------- 4951 4952AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4953 : BnAudioRecord(), 4954 mRecordTrack(recordTrack) 4955{ 4956} 4957 4958AudioFlinger::RecordHandle::~RecordHandle() { 4959 stop(); 4960} 4961 4962sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4963 return mRecordTrack->getCblk(); 4964} 4965 4966status_t AudioFlinger::RecordHandle::start(pid_t tid, int event, int triggerSession) { 4967 ALOGV("RecordHandle::start()"); 4968 return mRecordTrack->start(tid, (AudioSystem::sync_event_t)event, triggerSession); 4969} 4970 4971void AudioFlinger::RecordHandle::stop() { 4972 ALOGV("RecordHandle::stop()"); 4973 mRecordTrack->stop(); 4974} 4975 4976status_t AudioFlinger::RecordHandle::onTransact( 4977 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4978{ 4979 return BnAudioRecord::onTransact(code, data, reply, flags); 4980} 4981 4982// ---------------------------------------------------------------------------- 4983 4984AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4985 AudioStreamIn *input, 4986 uint32_t sampleRate, 4987 uint32_t channels, 4988 audio_io_handle_t id, 4989 uint32_t device) : 4990 ThreadBase(audioFlinger, id, device, RECORD), 4991 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4992 // mRsmpInIndex and mInputBytes set by readInputParameters() 4993 mReqChannelCount(popcount(channels)), 4994 mReqSampleRate(sampleRate) 4995 // mBytesRead is only meaningful while active, and so is cleared in start() 4996 // (but might be better to also clear here for dump?) 4997{ 4998 snprintf(mName, kNameLength, "AudioIn_%X", id); 4999 5000 readInputParameters(); 5001} 5002 5003 5004AudioFlinger::RecordThread::~RecordThread() 5005{ 5006 delete[] mRsmpInBuffer; 5007 delete mResampler; 5008 delete[] mRsmpOutBuffer; 5009} 5010 5011void AudioFlinger::RecordThread::onFirstRef() 5012{ 5013 run(mName, PRIORITY_URGENT_AUDIO); 5014} 5015 5016status_t AudioFlinger::RecordThread::readyToRun() 5017{ 5018 status_t status = initCheck(); 5019 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5020 return status; 5021} 5022 5023bool AudioFlinger::RecordThread::threadLoop() 5024{ 5025 AudioBufferProvider::Buffer buffer; 5026 sp<RecordTrack> activeTrack; 5027 Vector< sp<EffectChain> > effectChains; 5028 5029 nsecs_t lastWarning = 0; 5030 5031 acquireWakeLock(); 5032 5033 // start recording 5034 while (!exitPending()) { 5035 5036 processConfigEvents(); 5037 5038 { // scope for mLock 5039 Mutex::Autolock _l(mLock); 5040 checkForNewParameters_l(); 5041 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5042 if (!mStandby) { 5043 mInput->stream->common.standby(&mInput->stream->common); 5044 mStandby = true; 5045 } 5046 5047 if (exitPending()) break; 5048 5049 releaseWakeLock_l(); 5050 ALOGV("RecordThread: loop stopping"); 5051 // go to sleep 5052 mWaitWorkCV.wait(mLock); 5053 ALOGV("RecordThread: loop starting"); 5054 acquireWakeLock_l(); 5055 continue; 5056 } 5057 if (mActiveTrack != 0) { 5058 if (mActiveTrack->mState == TrackBase::PAUSING) { 5059 if (!mStandby) { 5060 mInput->stream->common.standby(&mInput->stream->common); 5061 mStandby = true; 5062 } 5063 mActiveTrack.clear(); 5064 mStartStopCond.broadcast(); 5065 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5066 if (mReqChannelCount != mActiveTrack->channelCount()) { 5067 mActiveTrack.clear(); 5068 mStartStopCond.broadcast(); 5069 } else if (mBytesRead != 0) { 5070 // record start succeeds only if first read from audio input 5071 // succeeds 5072 if (mBytesRead > 0) { 5073 mActiveTrack->mState = TrackBase::ACTIVE; 5074 } else { 5075 mActiveTrack.clear(); 5076 } 5077 mStartStopCond.broadcast(); 5078 } 5079 mStandby = false; 5080 } 5081 } 5082 lockEffectChains_l(effectChains); 5083 } 5084 5085 if (mActiveTrack != 0) { 5086 if (mActiveTrack->mState != TrackBase::ACTIVE && 5087 mActiveTrack->mState != TrackBase::RESUMING) { 5088 unlockEffectChains(effectChains); 5089 usleep(kRecordThreadSleepUs); 5090 continue; 5091 } 5092 for (size_t i = 0; i < effectChains.size(); i ++) { 5093 effectChains[i]->process_l(); 5094 } 5095 5096 buffer.frameCount = mFrameCount; 5097 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5098 size_t framesOut = buffer.frameCount; 5099 if (mResampler == NULL) { 5100 // no resampling 5101 while (framesOut) { 5102 size_t framesIn = mFrameCount - mRsmpInIndex; 5103 if (framesIn) { 5104 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5105 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5106 if (framesIn > framesOut) 5107 framesIn = framesOut; 5108 mRsmpInIndex += framesIn; 5109 framesOut -= framesIn; 5110 if ((int)mChannelCount == mReqChannelCount || 5111 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5112 memcpy(dst, src, framesIn * mFrameSize); 5113 } else { 5114 int16_t *src16 = (int16_t *)src; 5115 int16_t *dst16 = (int16_t *)dst; 5116 if (mChannelCount == 1) { 5117 while (framesIn--) { 5118 *dst16++ = *src16; 5119 *dst16++ = *src16++; 5120 } 5121 } else { 5122 while (framesIn--) { 5123 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5124 src16 += 2; 5125 } 5126 } 5127 } 5128 } 5129 if (framesOut && mFrameCount == mRsmpInIndex) { 5130 if (framesOut == mFrameCount && 5131 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5132 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5133 framesOut = 0; 5134 } else { 5135 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5136 mRsmpInIndex = 0; 5137 } 5138 if (mBytesRead < 0) { 5139 ALOGE("Error reading audio input"); 5140 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5141 // Force input into standby so that it tries to 5142 // recover at next read attempt 5143 mInput->stream->common.standby(&mInput->stream->common); 5144 usleep(kRecordThreadSleepUs); 5145 } 5146 mRsmpInIndex = mFrameCount; 5147 framesOut = 0; 5148 buffer.frameCount = 0; 5149 } 5150 } 5151 } 5152 } else { 5153 // resampling 5154 5155 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5156 // alter output frame count as if we were expecting stereo samples 5157 if (mChannelCount == 1 && mReqChannelCount == 1) { 5158 framesOut >>= 1; 5159 } 5160 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5161 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5162 // are 32 bit aligned which should be always true. 5163 if (mChannelCount == 2 && mReqChannelCount == 1) { 5164 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5165 // the resampler always outputs stereo samples: do post stereo to mono conversion 5166 int16_t *src = (int16_t *)mRsmpOutBuffer; 5167 int16_t *dst = buffer.i16; 5168 while (framesOut--) { 5169 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5170 src += 2; 5171 } 5172 } else { 5173 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5174 } 5175 5176 } 5177 if (mFramestoDrop == 0) { 5178 mActiveTrack->releaseBuffer(&buffer); 5179 } else { 5180 if (mFramestoDrop > 0) { 5181 mFramestoDrop -= buffer.frameCount; 5182 if (mFramestoDrop < 0) { 5183 mFramestoDrop = 0; 5184 } 5185 } 5186 } 5187 mActiveTrack->overflow(); 5188 } 5189 // client isn't retrieving buffers fast enough 5190 else { 5191 if (!mActiveTrack->setOverflow()) { 5192 nsecs_t now = systemTime(); 5193 if ((now - lastWarning) > kWarningThrottleNs) { 5194 ALOGW("RecordThread: buffer overflow"); 5195 lastWarning = now; 5196 } 5197 } 5198 // Release the processor for a while before asking for a new buffer. 5199 // This will give the application more chance to read from the buffer and 5200 // clear the overflow. 5201 usleep(kRecordThreadSleepUs); 5202 } 5203 } 5204 // enable changes in effect chain 5205 unlockEffectChains(effectChains); 5206 effectChains.clear(); 5207 } 5208 5209 if (!mStandby) { 5210 mInput->stream->common.standby(&mInput->stream->common); 5211 } 5212 mActiveTrack.clear(); 5213 5214 mStartStopCond.broadcast(); 5215 5216 releaseWakeLock(); 5217 5218 ALOGV("RecordThread %p exiting", this); 5219 return false; 5220} 5221 5222 5223sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5224 const sp<AudioFlinger::Client>& client, 5225 uint32_t sampleRate, 5226 audio_format_t format, 5227 int channelMask, 5228 int frameCount, 5229 int sessionId, 5230 status_t *status) 5231{ 5232 sp<RecordTrack> track; 5233 status_t lStatus; 5234 5235 lStatus = initCheck(); 5236 if (lStatus != NO_ERROR) { 5237 ALOGE("Audio driver not initialized."); 5238 goto Exit; 5239 } 5240 5241 { // scope for mLock 5242 Mutex::Autolock _l(mLock); 5243 5244 track = new RecordTrack(this, client, sampleRate, 5245 format, channelMask, frameCount, sessionId); 5246 5247 if (track->getCblk() == 0) { 5248 lStatus = NO_MEMORY; 5249 goto Exit; 5250 } 5251 5252 mTrack = track.get(); 5253 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5254 bool suspend = audio_is_bluetooth_sco_device( 5255 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5256 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5257 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5258 } 5259 lStatus = NO_ERROR; 5260 5261Exit: 5262 if (status) { 5263 *status = lStatus; 5264 } 5265 return track; 5266} 5267 5268status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5269 pid_t tid, AudioSystem::sync_event_t event, 5270 int triggerSession) 5271{ 5272 ALOGV("RecordThread::start tid=%d, event %d, triggerSession %d", tid, event, triggerSession); 5273 sp<ThreadBase> strongMe = this; 5274 status_t status = NO_ERROR; 5275 5276 if (event == AudioSystem::SYNC_EVENT_NONE) { 5277 mSyncStartEvent.clear(); 5278 mFramestoDrop = 0; 5279 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5280 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5281 triggerSession, 5282 recordTrack->sessionId(), 5283 syncStartEventCallback, 5284 this); 5285 mFramestoDrop = -1; 5286 } 5287 5288 { 5289 AutoMutex lock(mLock); 5290 if (mActiveTrack != 0) { 5291 if (recordTrack != mActiveTrack.get()) { 5292 status = -EBUSY; 5293 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5294 mActiveTrack->mState = TrackBase::ACTIVE; 5295 } 5296 return status; 5297 } 5298 5299 recordTrack->mState = TrackBase::IDLE; 5300 mActiveTrack = recordTrack; 5301 mLock.unlock(); 5302 status_t status = AudioSystem::startInput(mId); 5303 mLock.lock(); 5304 if (status != NO_ERROR) { 5305 mActiveTrack.clear(); 5306 clearSyncStartEvent(); 5307 return status; 5308 } 5309 mRsmpInIndex = mFrameCount; 5310 mBytesRead = 0; 5311 if (mResampler != NULL) { 5312 mResampler->reset(); 5313 } 5314 mActiveTrack->mState = TrackBase::RESUMING; 5315 // signal thread to start 5316 ALOGV("Signal record thread"); 5317 mWaitWorkCV.signal(); 5318 // do not wait for mStartStopCond if exiting 5319 if (exitPending()) { 5320 mActiveTrack.clear(); 5321 status = INVALID_OPERATION; 5322 goto startError; 5323 } 5324 mStartStopCond.wait(mLock); 5325 if (mActiveTrack == 0) { 5326 ALOGV("Record failed to start"); 5327 status = BAD_VALUE; 5328 goto startError; 5329 } 5330 ALOGV("Record started OK"); 5331 return status; 5332 } 5333startError: 5334 AudioSystem::stopInput(mId); 5335 clearSyncStartEvent(); 5336 return status; 5337} 5338 5339void AudioFlinger::RecordThread::clearSyncStartEvent() 5340{ 5341 if (mSyncStartEvent != 0) { 5342 mSyncStartEvent->cancel(); 5343 } 5344 mSyncStartEvent.clear(); 5345} 5346 5347void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5348{ 5349 sp<SyncEvent> strongEvent = event.promote(); 5350 5351 if (strongEvent != 0) { 5352 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5353 me->handleSyncStartEvent(strongEvent); 5354 } 5355} 5356 5357void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5358{ 5359 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5360 mActiveTrack.get(), 5361 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5362 event->listenerSession()); 5363 5364 if (mActiveTrack != 0 && 5365 event == mSyncStartEvent) { 5366 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5367 // from audio HAL 5368 mFramestoDrop = mFrameCount * 2; 5369 mSyncStartEvent.clear(); 5370 } 5371} 5372 5373void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5374 ALOGV("RecordThread::stop"); 5375 sp<ThreadBase> strongMe = this; 5376 { 5377 AutoMutex lock(mLock); 5378 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5379 mActiveTrack->mState = TrackBase::PAUSING; 5380 // do not wait for mStartStopCond if exiting 5381 if (exitPending()) { 5382 return; 5383 } 5384 mStartStopCond.wait(mLock); 5385 // if we have been restarted, recordTrack == mActiveTrack.get() here 5386 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5387 mLock.unlock(); 5388 AudioSystem::stopInput(mId); 5389 mLock.lock(); 5390 ALOGV("Record stopped OK"); 5391 } 5392 } 5393 } 5394} 5395 5396bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5397{ 5398 return false; 5399} 5400 5401status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5402{ 5403 if (!isValidSyncEvent(event)) { 5404 return BAD_VALUE; 5405 } 5406 5407 Mutex::Autolock _l(mLock); 5408 5409 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5410 mTrack->setSyncEvent(event); 5411 return NO_ERROR; 5412 } 5413 return NAME_NOT_FOUND; 5414} 5415 5416status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5417{ 5418 const size_t SIZE = 256; 5419 char buffer[SIZE]; 5420 String8 result; 5421 5422 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5423 result.append(buffer); 5424 5425 if (mActiveTrack != 0) { 5426 result.append("Active Track:\n"); 5427 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5428 mActiveTrack->dump(buffer, SIZE); 5429 result.append(buffer); 5430 5431 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5432 result.append(buffer); 5433 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5434 result.append(buffer); 5435 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5436 result.append(buffer); 5437 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5438 result.append(buffer); 5439 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5440 result.append(buffer); 5441 5442 5443 } else { 5444 result.append("No record client\n"); 5445 } 5446 write(fd, result.string(), result.size()); 5447 5448 dumpBase(fd, args); 5449 dumpEffectChains(fd, args); 5450 5451 return NO_ERROR; 5452} 5453 5454// AudioBufferProvider interface 5455status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5456{ 5457 size_t framesReq = buffer->frameCount; 5458 size_t framesReady = mFrameCount - mRsmpInIndex; 5459 int channelCount; 5460 5461 if (framesReady == 0) { 5462 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5463 if (mBytesRead < 0) { 5464 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5465 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5466 // Force input into standby so that it tries to 5467 // recover at next read attempt 5468 mInput->stream->common.standby(&mInput->stream->common); 5469 usleep(kRecordThreadSleepUs); 5470 } 5471 buffer->raw = NULL; 5472 buffer->frameCount = 0; 5473 return NOT_ENOUGH_DATA; 5474 } 5475 mRsmpInIndex = 0; 5476 framesReady = mFrameCount; 5477 } 5478 5479 if (framesReq > framesReady) { 5480 framesReq = framesReady; 5481 } 5482 5483 if (mChannelCount == 1 && mReqChannelCount == 2) { 5484 channelCount = 1; 5485 } else { 5486 channelCount = 2; 5487 } 5488 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5489 buffer->frameCount = framesReq; 5490 return NO_ERROR; 5491} 5492 5493// AudioBufferProvider interface 5494void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5495{ 5496 mRsmpInIndex += buffer->frameCount; 5497 buffer->frameCount = 0; 5498} 5499 5500bool AudioFlinger::RecordThread::checkForNewParameters_l() 5501{ 5502 bool reconfig = false; 5503 5504 while (!mNewParameters.isEmpty()) { 5505 status_t status = NO_ERROR; 5506 String8 keyValuePair = mNewParameters[0]; 5507 AudioParameter param = AudioParameter(keyValuePair); 5508 int value; 5509 audio_format_t reqFormat = mFormat; 5510 int reqSamplingRate = mReqSampleRate; 5511 int reqChannelCount = mReqChannelCount; 5512 5513 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5514 reqSamplingRate = value; 5515 reconfig = true; 5516 } 5517 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5518 reqFormat = (audio_format_t) value; 5519 reconfig = true; 5520 } 5521 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5522 reqChannelCount = popcount(value); 5523 reconfig = true; 5524 } 5525 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5526 // do not accept frame count changes if tracks are open as the track buffer 5527 // size depends on frame count and correct behavior would not be guaranteed 5528 // if frame count is changed after track creation 5529 if (mActiveTrack != 0) { 5530 status = INVALID_OPERATION; 5531 } else { 5532 reconfig = true; 5533 } 5534 } 5535 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5536 // forward device change to effects that have requested to be 5537 // aware of attached audio device. 5538 for (size_t i = 0; i < mEffectChains.size(); i++) { 5539 mEffectChains[i]->setDevice_l(value); 5540 } 5541 // store input device and output device but do not forward output device to audio HAL. 5542 // Note that status is ignored by the caller for output device 5543 // (see AudioFlinger::setParameters() 5544 if (value & AUDIO_DEVICE_OUT_ALL) { 5545 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5546 status = BAD_VALUE; 5547 } else { 5548 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5549 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5550 if (mTrack != NULL) { 5551 bool suspend = audio_is_bluetooth_sco_device( 5552 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5553 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5554 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5555 } 5556 } 5557 mDevice |= (uint32_t)value; 5558 } 5559 if (status == NO_ERROR) { 5560 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5561 if (status == INVALID_OPERATION) { 5562 mInput->stream->common.standby(&mInput->stream->common); 5563 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5564 keyValuePair.string()); 5565 } 5566 if (reconfig) { 5567 if (status == BAD_VALUE && 5568 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5569 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5570 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5571 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5572 (reqChannelCount <= FCC_2)) { 5573 status = NO_ERROR; 5574 } 5575 if (status == NO_ERROR) { 5576 readInputParameters(); 5577 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5578 } 5579 } 5580 } 5581 5582 mNewParameters.removeAt(0); 5583 5584 mParamStatus = status; 5585 mParamCond.signal(); 5586 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5587 // already timed out waiting for the status and will never signal the condition. 5588 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5589 } 5590 return reconfig; 5591} 5592 5593String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5594{ 5595 char *s; 5596 String8 out_s8 = String8(); 5597 5598 Mutex::Autolock _l(mLock); 5599 if (initCheck() != NO_ERROR) { 5600 return out_s8; 5601 } 5602 5603 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5604 out_s8 = String8(s); 5605 free(s); 5606 return out_s8; 5607} 5608 5609void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5610 AudioSystem::OutputDescriptor desc; 5611 void *param2 = NULL; 5612 5613 switch (event) { 5614 case AudioSystem::INPUT_OPENED: 5615 case AudioSystem::INPUT_CONFIG_CHANGED: 5616 desc.channels = mChannelMask; 5617 desc.samplingRate = mSampleRate; 5618 desc.format = mFormat; 5619 desc.frameCount = mFrameCount; 5620 desc.latency = 0; 5621 param2 = &desc; 5622 break; 5623 5624 case AudioSystem::INPUT_CLOSED: 5625 default: 5626 break; 5627 } 5628 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5629} 5630 5631void AudioFlinger::RecordThread::readInputParameters() 5632{ 5633 delete mRsmpInBuffer; 5634 // mRsmpInBuffer is always assigned a new[] below 5635 delete mRsmpOutBuffer; 5636 mRsmpOutBuffer = NULL; 5637 delete mResampler; 5638 mResampler = NULL; 5639 5640 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5641 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5642 mChannelCount = (uint16_t)popcount(mChannelMask); 5643 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5644 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5645 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5646 mFrameCount = mInputBytes / mFrameSize; 5647 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5648 5649 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5650 { 5651 int channelCount; 5652 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5653 // stereo to mono post process as the resampler always outputs stereo. 5654 if (mChannelCount == 1 && mReqChannelCount == 2) { 5655 channelCount = 1; 5656 } else { 5657 channelCount = 2; 5658 } 5659 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5660 mResampler->setSampleRate(mSampleRate); 5661 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5662 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5663 5664 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5665 if (mChannelCount == 1 && mReqChannelCount == 1) { 5666 mFrameCount >>= 1; 5667 } 5668 5669 } 5670 mRsmpInIndex = mFrameCount; 5671} 5672 5673unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5674{ 5675 Mutex::Autolock _l(mLock); 5676 if (initCheck() != NO_ERROR) { 5677 return 0; 5678 } 5679 5680 return mInput->stream->get_input_frames_lost(mInput->stream); 5681} 5682 5683uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5684{ 5685 Mutex::Autolock _l(mLock); 5686 uint32_t result = 0; 5687 if (getEffectChain_l(sessionId) != 0) { 5688 result = EFFECT_SESSION; 5689 } 5690 5691 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5692 result |= TRACK_SESSION; 5693 } 5694 5695 return result; 5696} 5697 5698AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5699{ 5700 Mutex::Autolock _l(mLock); 5701 return mTrack; 5702} 5703 5704AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5705{ 5706 Mutex::Autolock _l(mLock); 5707 return mInput; 5708} 5709 5710AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5711{ 5712 Mutex::Autolock _l(mLock); 5713 AudioStreamIn *input = mInput; 5714 mInput = NULL; 5715 return input; 5716} 5717 5718// this method must always be called either with ThreadBase mLock held or inside the thread loop 5719audio_stream_t* AudioFlinger::RecordThread::stream() const 5720{ 5721 if (mInput == NULL) { 5722 return NULL; 5723 } 5724 return &mInput->stream->common; 5725} 5726 5727 5728// ---------------------------------------------------------------------------- 5729 5730audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 5731{ 5732 if (!settingsAllowed()) { 5733 return 0; 5734 } 5735 Mutex::Autolock _l(mLock); 5736 return loadHwModule_l(name); 5737} 5738 5739// loadHwModule_l() must be called with AudioFlinger::mLock held 5740audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 5741{ 5742 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5743 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 5744 ALOGW("loadHwModule() module %s already loaded", name); 5745 return mAudioHwDevs.keyAt(i); 5746 } 5747 } 5748 5749 audio_hw_device_t *dev; 5750 5751 int rc = load_audio_interface(name, &dev); 5752 if (rc) { 5753 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 5754 return 0; 5755 } 5756 5757 mHardwareStatus = AUDIO_HW_INIT; 5758 rc = dev->init_check(dev); 5759 mHardwareStatus = AUDIO_HW_IDLE; 5760 if (rc) { 5761 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 5762 return 0; 5763 } 5764 5765 if ((mMasterVolumeSupportLvl != MVS_NONE) && 5766 (NULL != dev->set_master_volume)) { 5767 AutoMutex lock(mHardwareLock); 5768 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5769 dev->set_master_volume(dev, mMasterVolume); 5770 mHardwareStatus = AUDIO_HW_IDLE; 5771 } 5772 5773 audio_module_handle_t handle = nextUniqueId(); 5774 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 5775 5776 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 5777 name, dev->common.module->name, dev->common.module->id, handle); 5778 5779 return handle; 5780 5781} 5782 5783audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 5784 audio_devices_t *pDevices, 5785 uint32_t *pSamplingRate, 5786 audio_format_t *pFormat, 5787 audio_channel_mask_t *pChannelMask, 5788 uint32_t *pLatencyMs, 5789 audio_output_flags_t flags) 5790{ 5791 status_t status; 5792 PlaybackThread *thread = NULL; 5793 struct audio_config config = { 5794 sample_rate: pSamplingRate ? *pSamplingRate : 0, 5795 channel_mask: pChannelMask ? *pChannelMask : 0, 5796 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 5797 }; 5798 audio_stream_out_t *outStream = NULL; 5799 audio_hw_device_t *outHwDev; 5800 5801 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5802 module, 5803 (pDevices != NULL) ? (int)*pDevices : 0, 5804 config.sample_rate, 5805 config.format, 5806 config.channel_mask, 5807 flags); 5808 5809 if (pDevices == NULL || *pDevices == 0) { 5810 return 0; 5811 } 5812 5813 Mutex::Autolock _l(mLock); 5814 5815 outHwDev = findSuitableHwDev_l(module, *pDevices); 5816 if (outHwDev == NULL) 5817 return 0; 5818 5819 audio_io_handle_t id = nextUniqueId(); 5820 5821 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5822 5823 status = outHwDev->open_output_stream(outHwDev, 5824 id, 5825 *pDevices, 5826 (audio_output_flags_t)flags, 5827 &config, 5828 &outStream); 5829 5830 mHardwareStatus = AUDIO_HW_IDLE; 5831 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5832 outStream, 5833 config.sample_rate, 5834 config.format, 5835 config.channel_mask, 5836 status); 5837 5838 if (status == NO_ERROR && outStream != NULL) { 5839 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5840 5841 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 5842 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 5843 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 5844 thread = new DirectOutputThread(this, output, id, *pDevices); 5845 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5846 } else { 5847 thread = new MixerThread(this, output, id, *pDevices); 5848 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5849 } 5850 mPlaybackThreads.add(id, thread); 5851 5852 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 5853 if (pFormat != NULL) *pFormat = config.format; 5854 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 5855 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5856 5857 // notify client processes of the new output creation 5858 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5859 5860 // the first primary output opened designates the primary hw device 5861 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 5862 ALOGI("Using module %d has the primary audio interface", module); 5863 mPrimaryHardwareDev = outHwDev; 5864 5865 AutoMutex lock(mHardwareLock); 5866 mHardwareStatus = AUDIO_HW_SET_MODE; 5867 outHwDev->set_mode(outHwDev, mMode); 5868 5869 // Determine the level of master volume support the primary audio HAL has, 5870 // and set the initial master volume at the same time. 5871 float initialVolume = 1.0; 5872 mMasterVolumeSupportLvl = MVS_NONE; 5873 5874 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 5875 if ((NULL != outHwDev->get_master_volume) && 5876 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 5877 mMasterVolumeSupportLvl = MVS_FULL; 5878 } else { 5879 mMasterVolumeSupportLvl = MVS_SETONLY; 5880 initialVolume = 1.0; 5881 } 5882 5883 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5884 if ((NULL == outHwDev->set_master_volume) || 5885 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 5886 mMasterVolumeSupportLvl = MVS_NONE; 5887 } 5888 // now that we have a primary device, initialize master volume on other devices 5889 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5890 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 5891 5892 if ((dev != mPrimaryHardwareDev) && 5893 (NULL != dev->set_master_volume)) { 5894 dev->set_master_volume(dev, initialVolume); 5895 } 5896 } 5897 mHardwareStatus = AUDIO_HW_IDLE; 5898 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 5899 ? initialVolume 5900 : 1.0; 5901 mMasterVolume = initialVolume; 5902 } 5903 return id; 5904 } 5905 5906 return 0; 5907} 5908 5909audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5910 audio_io_handle_t output2) 5911{ 5912 Mutex::Autolock _l(mLock); 5913 MixerThread *thread1 = checkMixerThread_l(output1); 5914 MixerThread *thread2 = checkMixerThread_l(output2); 5915 5916 if (thread1 == NULL || thread2 == NULL) { 5917 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5918 return 0; 5919 } 5920 5921 audio_io_handle_t id = nextUniqueId(); 5922 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5923 thread->addOutputTrack(thread2); 5924 mPlaybackThreads.add(id, thread); 5925 // notify client processes of the new output creation 5926 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5927 return id; 5928} 5929 5930status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5931{ 5932 // keep strong reference on the playback thread so that 5933 // it is not destroyed while exit() is executed 5934 sp<PlaybackThread> thread; 5935 { 5936 Mutex::Autolock _l(mLock); 5937 thread = checkPlaybackThread_l(output); 5938 if (thread == NULL) { 5939 return BAD_VALUE; 5940 } 5941 5942 ALOGV("closeOutput() %d", output); 5943 5944 if (thread->type() == ThreadBase::MIXER) { 5945 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5946 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5947 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5948 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5949 } 5950 } 5951 } 5952 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5953 mPlaybackThreads.removeItem(output); 5954 } 5955 thread->exit(); 5956 // The thread entity (active unit of execution) is no longer running here, 5957 // but the ThreadBase container still exists. 5958 5959 if (thread->type() != ThreadBase::DUPLICATING) { 5960 AudioStreamOut *out = thread->clearOutput(); 5961 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5962 // from now on thread->mOutput is NULL 5963 out->hwDev->close_output_stream(out->hwDev, out->stream); 5964 delete out; 5965 } 5966 return NO_ERROR; 5967} 5968 5969status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5970{ 5971 Mutex::Autolock _l(mLock); 5972 PlaybackThread *thread = checkPlaybackThread_l(output); 5973 5974 if (thread == NULL) { 5975 return BAD_VALUE; 5976 } 5977 5978 ALOGV("suspendOutput() %d", output); 5979 thread->suspend(); 5980 5981 return NO_ERROR; 5982} 5983 5984status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5985{ 5986 Mutex::Autolock _l(mLock); 5987 PlaybackThread *thread = checkPlaybackThread_l(output); 5988 5989 if (thread == NULL) { 5990 return BAD_VALUE; 5991 } 5992 5993 ALOGV("restoreOutput() %d", output); 5994 5995 thread->restore(); 5996 5997 return NO_ERROR; 5998} 5999 6000audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6001 audio_devices_t *pDevices, 6002 uint32_t *pSamplingRate, 6003 audio_format_t *pFormat, 6004 uint32_t *pChannelMask) 6005{ 6006 status_t status; 6007 RecordThread *thread = NULL; 6008 struct audio_config config = { 6009 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6010 channel_mask: pChannelMask ? *pChannelMask : 0, 6011 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6012 }; 6013 uint32_t reqSamplingRate = config.sample_rate; 6014 audio_format_t reqFormat = config.format; 6015 audio_channel_mask_t reqChannels = config.channel_mask; 6016 audio_stream_in_t *inStream = NULL; 6017 audio_hw_device_t *inHwDev; 6018 6019 if (pDevices == NULL || *pDevices == 0) { 6020 return 0; 6021 } 6022 6023 Mutex::Autolock _l(mLock); 6024 6025 inHwDev = findSuitableHwDev_l(module, *pDevices); 6026 if (inHwDev == NULL) 6027 return 0; 6028 6029 audio_io_handle_t id = nextUniqueId(); 6030 6031 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6032 &inStream); 6033 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6034 inStream, 6035 config.sample_rate, 6036 config.format, 6037 config.channel_mask, 6038 status); 6039 6040 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6041 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6042 // or stereo to mono conversions on 16 bit PCM inputs. 6043 if (status == BAD_VALUE && 6044 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6045 (config.sample_rate <= 2 * reqSamplingRate) && 6046 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6047 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6048 inStream = NULL; 6049 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6050 } 6051 6052 if (status == NO_ERROR && inStream != NULL) { 6053 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6054 6055 // Start record thread 6056 // RecorThread require both input and output device indication to forward to audio 6057 // pre processing modules 6058 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6059 thread = new RecordThread(this, 6060 input, 6061 reqSamplingRate, 6062 reqChannels, 6063 id, 6064 device); 6065 mRecordThreads.add(id, thread); 6066 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6067 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6068 if (pFormat != NULL) *pFormat = config.format; 6069 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6070 6071 input->stream->common.standby(&input->stream->common); 6072 6073 // notify client processes of the new input creation 6074 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6075 return id; 6076 } 6077 6078 return 0; 6079} 6080 6081status_t AudioFlinger::closeInput(audio_io_handle_t input) 6082{ 6083 // keep strong reference on the record thread so that 6084 // it is not destroyed while exit() is executed 6085 sp<RecordThread> thread; 6086 { 6087 Mutex::Autolock _l(mLock); 6088 thread = checkRecordThread_l(input); 6089 if (thread == NULL) { 6090 return BAD_VALUE; 6091 } 6092 6093 ALOGV("closeInput() %d", input); 6094 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6095 mRecordThreads.removeItem(input); 6096 } 6097 thread->exit(); 6098 // The thread entity (active unit of execution) is no longer running here, 6099 // but the ThreadBase container still exists. 6100 6101 AudioStreamIn *in = thread->clearInput(); 6102 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6103 // from now on thread->mInput is NULL 6104 in->hwDev->close_input_stream(in->hwDev, in->stream); 6105 delete in; 6106 6107 return NO_ERROR; 6108} 6109 6110status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6111{ 6112 Mutex::Autolock _l(mLock); 6113 MixerThread *dstThread = checkMixerThread_l(output); 6114 if (dstThread == NULL) { 6115 ALOGW("setStreamOutput() bad output id %d", output); 6116 return BAD_VALUE; 6117 } 6118 6119 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6120 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6121 6122 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6123 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6124 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6125 MixerThread *srcThread = (MixerThread *)thread; 6126 srcThread->invalidateTracks(stream); 6127 } 6128 } 6129 6130 return NO_ERROR; 6131} 6132 6133 6134int AudioFlinger::newAudioSessionId() 6135{ 6136 return nextUniqueId(); 6137} 6138 6139void AudioFlinger::acquireAudioSessionId(int audioSession) 6140{ 6141 Mutex::Autolock _l(mLock); 6142 pid_t caller = IPCThreadState::self()->getCallingPid(); 6143 ALOGV("acquiring %d from %d", audioSession, caller); 6144 size_t num = mAudioSessionRefs.size(); 6145 for (size_t i = 0; i< num; i++) { 6146 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6147 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6148 ref->mCnt++; 6149 ALOGV(" incremented refcount to %d", ref->mCnt); 6150 return; 6151 } 6152 } 6153 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6154 ALOGV(" added new entry for %d", audioSession); 6155} 6156 6157void AudioFlinger::releaseAudioSessionId(int audioSession) 6158{ 6159 Mutex::Autolock _l(mLock); 6160 pid_t caller = IPCThreadState::self()->getCallingPid(); 6161 ALOGV("releasing %d from %d", audioSession, caller); 6162 size_t num = mAudioSessionRefs.size(); 6163 for (size_t i = 0; i< num; i++) { 6164 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6165 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6166 ref->mCnt--; 6167 ALOGV(" decremented refcount to %d", ref->mCnt); 6168 if (ref->mCnt == 0) { 6169 mAudioSessionRefs.removeAt(i); 6170 delete ref; 6171 purgeStaleEffects_l(); 6172 } 6173 return; 6174 } 6175 } 6176 ALOGW("session id %d not found for pid %d", audioSession, caller); 6177} 6178 6179void AudioFlinger::purgeStaleEffects_l() { 6180 6181 ALOGV("purging stale effects"); 6182 6183 Vector< sp<EffectChain> > chains; 6184 6185 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6186 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6187 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6188 sp<EffectChain> ec = t->mEffectChains[j]; 6189 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6190 chains.push(ec); 6191 } 6192 } 6193 } 6194 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6195 sp<RecordThread> t = mRecordThreads.valueAt(i); 6196 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6197 sp<EffectChain> ec = t->mEffectChains[j]; 6198 chains.push(ec); 6199 } 6200 } 6201 6202 for (size_t i = 0; i < chains.size(); i++) { 6203 sp<EffectChain> ec = chains[i]; 6204 int sessionid = ec->sessionId(); 6205 sp<ThreadBase> t = ec->mThread.promote(); 6206 if (t == 0) { 6207 continue; 6208 } 6209 size_t numsessionrefs = mAudioSessionRefs.size(); 6210 bool found = false; 6211 for (size_t k = 0; k < numsessionrefs; k++) { 6212 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6213 if (ref->mSessionid == sessionid) { 6214 ALOGV(" session %d still exists for %d with %d refs", 6215 sessionid, ref->mPid, ref->mCnt); 6216 found = true; 6217 break; 6218 } 6219 } 6220 if (!found) { 6221 // remove all effects from the chain 6222 while (ec->mEffects.size()) { 6223 sp<EffectModule> effect = ec->mEffects[0]; 6224 effect->unPin(); 6225 Mutex::Autolock _l (t->mLock); 6226 t->removeEffect_l(effect); 6227 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6228 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6229 if (handle != 0) { 6230 handle->mEffect.clear(); 6231 if (handle->mHasControl && handle->mEnabled) { 6232 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6233 } 6234 } 6235 } 6236 AudioSystem::unregisterEffect(effect->id()); 6237 } 6238 } 6239 } 6240 return; 6241} 6242 6243// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6244AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6245{ 6246 return mPlaybackThreads.valueFor(output).get(); 6247} 6248 6249// checkMixerThread_l() must be called with AudioFlinger::mLock held 6250AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6251{ 6252 PlaybackThread *thread = checkPlaybackThread_l(output); 6253 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6254} 6255 6256// checkRecordThread_l() must be called with AudioFlinger::mLock held 6257AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6258{ 6259 return mRecordThreads.valueFor(input).get(); 6260} 6261 6262uint32_t AudioFlinger::nextUniqueId() 6263{ 6264 return android_atomic_inc(&mNextUniqueId); 6265} 6266 6267AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6268{ 6269 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6270 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6271 AudioStreamOut *output = thread->getOutput(); 6272 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6273 return thread; 6274 } 6275 } 6276 return NULL; 6277} 6278 6279uint32_t AudioFlinger::primaryOutputDevice_l() const 6280{ 6281 PlaybackThread *thread = primaryPlaybackThread_l(); 6282 6283 if (thread == NULL) { 6284 return 0; 6285 } 6286 6287 return thread->device(); 6288} 6289 6290sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6291 int triggerSession, 6292 int listenerSession, 6293 sync_event_callback_t callBack, 6294 void *cookie) 6295{ 6296 Mutex::Autolock _l(mLock); 6297 6298 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6299 status_t playStatus = NAME_NOT_FOUND; 6300 status_t recStatus = NAME_NOT_FOUND; 6301 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6302 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6303 if (playStatus == NO_ERROR) { 6304 return event; 6305 } 6306 } 6307 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6308 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6309 if (recStatus == NO_ERROR) { 6310 return event; 6311 } 6312 } 6313 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6314 mPendingSyncEvents.add(event); 6315 } else { 6316 ALOGV("createSyncEvent() invalid event %d", event->type()); 6317 event.clear(); 6318 } 6319 return event; 6320} 6321 6322// ---------------------------------------------------------------------------- 6323// Effect management 6324// ---------------------------------------------------------------------------- 6325 6326 6327status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6328{ 6329 Mutex::Autolock _l(mLock); 6330 return EffectQueryNumberEffects(numEffects); 6331} 6332 6333status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6334{ 6335 Mutex::Autolock _l(mLock); 6336 return EffectQueryEffect(index, descriptor); 6337} 6338 6339status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6340 effect_descriptor_t *descriptor) const 6341{ 6342 Mutex::Autolock _l(mLock); 6343 return EffectGetDescriptor(pUuid, descriptor); 6344} 6345 6346 6347sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6348 effect_descriptor_t *pDesc, 6349 const sp<IEffectClient>& effectClient, 6350 int32_t priority, 6351 audio_io_handle_t io, 6352 int sessionId, 6353 status_t *status, 6354 int *id, 6355 int *enabled) 6356{ 6357 status_t lStatus = NO_ERROR; 6358 sp<EffectHandle> handle; 6359 effect_descriptor_t desc; 6360 6361 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6362 pid, effectClient.get(), priority, sessionId, io); 6363 6364 if (pDesc == NULL) { 6365 lStatus = BAD_VALUE; 6366 goto Exit; 6367 } 6368 6369 // check audio settings permission for global effects 6370 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6371 lStatus = PERMISSION_DENIED; 6372 goto Exit; 6373 } 6374 6375 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6376 // that can only be created by audio policy manager (running in same process) 6377 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6378 lStatus = PERMISSION_DENIED; 6379 goto Exit; 6380 } 6381 6382 if (io == 0) { 6383 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6384 // output must be specified by AudioPolicyManager when using session 6385 // AUDIO_SESSION_OUTPUT_STAGE 6386 lStatus = BAD_VALUE; 6387 goto Exit; 6388 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6389 // if the output returned by getOutputForEffect() is removed before we lock the 6390 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6391 // and we will exit safely 6392 io = AudioSystem::getOutputForEffect(&desc); 6393 } 6394 } 6395 6396 { 6397 Mutex::Autolock _l(mLock); 6398 6399 6400 if (!EffectIsNullUuid(&pDesc->uuid)) { 6401 // if uuid is specified, request effect descriptor 6402 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6403 if (lStatus < 0) { 6404 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6405 goto Exit; 6406 } 6407 } else { 6408 // if uuid is not specified, look for an available implementation 6409 // of the required type in effect factory 6410 if (EffectIsNullUuid(&pDesc->type)) { 6411 ALOGW("createEffect() no effect type"); 6412 lStatus = BAD_VALUE; 6413 goto Exit; 6414 } 6415 uint32_t numEffects = 0; 6416 effect_descriptor_t d; 6417 d.flags = 0; // prevent compiler warning 6418 bool found = false; 6419 6420 lStatus = EffectQueryNumberEffects(&numEffects); 6421 if (lStatus < 0) { 6422 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6423 goto Exit; 6424 } 6425 for (uint32_t i = 0; i < numEffects; i++) { 6426 lStatus = EffectQueryEffect(i, &desc); 6427 if (lStatus < 0) { 6428 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6429 continue; 6430 } 6431 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6432 // If matching type found save effect descriptor. If the session is 6433 // 0 and the effect is not auxiliary, continue enumeration in case 6434 // an auxiliary version of this effect type is available 6435 found = true; 6436 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6437 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6438 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6439 break; 6440 } 6441 } 6442 } 6443 if (!found) { 6444 lStatus = BAD_VALUE; 6445 ALOGW("createEffect() effect not found"); 6446 goto Exit; 6447 } 6448 // For same effect type, chose auxiliary version over insert version if 6449 // connect to output mix (Compliance to OpenSL ES) 6450 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6451 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6452 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6453 } 6454 } 6455 6456 // Do not allow auxiliary effects on a session different from 0 (output mix) 6457 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6458 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6459 lStatus = INVALID_OPERATION; 6460 goto Exit; 6461 } 6462 6463 // check recording permission for visualizer 6464 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6465 !recordingAllowed()) { 6466 lStatus = PERMISSION_DENIED; 6467 goto Exit; 6468 } 6469 6470 // return effect descriptor 6471 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6472 6473 // If output is not specified try to find a matching audio session ID in one of the 6474 // output threads. 6475 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6476 // because of code checking output when entering the function. 6477 // Note: io is never 0 when creating an effect on an input 6478 if (io == 0) { 6479 // look for the thread where the specified audio session is present 6480 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6481 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6482 io = mPlaybackThreads.keyAt(i); 6483 break; 6484 } 6485 } 6486 if (io == 0) { 6487 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6488 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6489 io = mRecordThreads.keyAt(i); 6490 break; 6491 } 6492 } 6493 } 6494 // If no output thread contains the requested session ID, default to 6495 // first output. The effect chain will be moved to the correct output 6496 // thread when a track with the same session ID is created 6497 if (io == 0 && mPlaybackThreads.size()) { 6498 io = mPlaybackThreads.keyAt(0); 6499 } 6500 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6501 } 6502 ThreadBase *thread = checkRecordThread_l(io); 6503 if (thread == NULL) { 6504 thread = checkPlaybackThread_l(io); 6505 if (thread == NULL) { 6506 ALOGE("createEffect() unknown output thread"); 6507 lStatus = BAD_VALUE; 6508 goto Exit; 6509 } 6510 } 6511 6512 sp<Client> client = registerPid_l(pid); 6513 6514 // create effect on selected output thread 6515 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6516 &desc, enabled, &lStatus); 6517 if (handle != 0 && id != NULL) { 6518 *id = handle->id(); 6519 } 6520 } 6521 6522Exit: 6523 if (status != NULL) { 6524 *status = lStatus; 6525 } 6526 return handle; 6527} 6528 6529status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6530 audio_io_handle_t dstOutput) 6531{ 6532 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6533 sessionId, srcOutput, dstOutput); 6534 Mutex::Autolock _l(mLock); 6535 if (srcOutput == dstOutput) { 6536 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6537 return NO_ERROR; 6538 } 6539 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6540 if (srcThread == NULL) { 6541 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6542 return BAD_VALUE; 6543 } 6544 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6545 if (dstThread == NULL) { 6546 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6547 return BAD_VALUE; 6548 } 6549 6550 Mutex::Autolock _dl(dstThread->mLock); 6551 Mutex::Autolock _sl(srcThread->mLock); 6552 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6553 6554 return NO_ERROR; 6555} 6556 6557// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6558status_t AudioFlinger::moveEffectChain_l(int sessionId, 6559 AudioFlinger::PlaybackThread *srcThread, 6560 AudioFlinger::PlaybackThread *dstThread, 6561 bool reRegister) 6562{ 6563 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6564 sessionId, srcThread, dstThread); 6565 6566 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6567 if (chain == 0) { 6568 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6569 sessionId, srcThread); 6570 return INVALID_OPERATION; 6571 } 6572 6573 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6574 // so that a new chain is created with correct parameters when first effect is added. This is 6575 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6576 // removed. 6577 srcThread->removeEffectChain_l(chain); 6578 6579 // transfer all effects one by one so that new effect chain is created on new thread with 6580 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6581 audio_io_handle_t dstOutput = dstThread->id(); 6582 sp<EffectChain> dstChain; 6583 uint32_t strategy = 0; // prevent compiler warning 6584 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6585 while (effect != 0) { 6586 srcThread->removeEffect_l(effect); 6587 dstThread->addEffect_l(effect); 6588 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6589 if (effect->state() == EffectModule::ACTIVE || 6590 effect->state() == EffectModule::STOPPING) { 6591 effect->start(); 6592 } 6593 // if the move request is not received from audio policy manager, the effect must be 6594 // re-registered with the new strategy and output 6595 if (dstChain == 0) { 6596 dstChain = effect->chain().promote(); 6597 if (dstChain == 0) { 6598 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6599 srcThread->addEffect_l(effect); 6600 return NO_INIT; 6601 } 6602 strategy = dstChain->strategy(); 6603 } 6604 if (reRegister) { 6605 AudioSystem::unregisterEffect(effect->id()); 6606 AudioSystem::registerEffect(&effect->desc(), 6607 dstOutput, 6608 strategy, 6609 sessionId, 6610 effect->id()); 6611 } 6612 effect = chain->getEffectFromId_l(0); 6613 } 6614 6615 return NO_ERROR; 6616} 6617 6618 6619// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6620sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6621 const sp<AudioFlinger::Client>& client, 6622 const sp<IEffectClient>& effectClient, 6623 int32_t priority, 6624 int sessionId, 6625 effect_descriptor_t *desc, 6626 int *enabled, 6627 status_t *status 6628 ) 6629{ 6630 sp<EffectModule> effect; 6631 sp<EffectHandle> handle; 6632 status_t lStatus; 6633 sp<EffectChain> chain; 6634 bool chainCreated = false; 6635 bool effectCreated = false; 6636 bool effectRegistered = false; 6637 6638 lStatus = initCheck(); 6639 if (lStatus != NO_ERROR) { 6640 ALOGW("createEffect_l() Audio driver not initialized."); 6641 goto Exit; 6642 } 6643 6644 // Do not allow effects with session ID 0 on direct output or duplicating threads 6645 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6646 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6647 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6648 desc->name, sessionId); 6649 lStatus = BAD_VALUE; 6650 goto Exit; 6651 } 6652 // Only Pre processor effects are allowed on input threads and only on input threads 6653 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6654 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6655 desc->name, desc->flags, mType); 6656 lStatus = BAD_VALUE; 6657 goto Exit; 6658 } 6659 6660 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6661 6662 { // scope for mLock 6663 Mutex::Autolock _l(mLock); 6664 6665 // check for existing effect chain with the requested audio session 6666 chain = getEffectChain_l(sessionId); 6667 if (chain == 0) { 6668 // create a new chain for this session 6669 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6670 chain = new EffectChain(this, sessionId); 6671 addEffectChain_l(chain); 6672 chain->setStrategy(getStrategyForSession_l(sessionId)); 6673 chainCreated = true; 6674 } else { 6675 effect = chain->getEffectFromDesc_l(desc); 6676 } 6677 6678 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6679 6680 if (effect == 0) { 6681 int id = mAudioFlinger->nextUniqueId(); 6682 // Check CPU and memory usage 6683 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6684 if (lStatus != NO_ERROR) { 6685 goto Exit; 6686 } 6687 effectRegistered = true; 6688 // create a new effect module if none present in the chain 6689 effect = new EffectModule(this, chain, desc, id, sessionId); 6690 lStatus = effect->status(); 6691 if (lStatus != NO_ERROR) { 6692 goto Exit; 6693 } 6694 lStatus = chain->addEffect_l(effect); 6695 if (lStatus != NO_ERROR) { 6696 goto Exit; 6697 } 6698 effectCreated = true; 6699 6700 effect->setDevice(mDevice); 6701 effect->setMode(mAudioFlinger->getMode()); 6702 } 6703 // create effect handle and connect it to effect module 6704 handle = new EffectHandle(effect, client, effectClient, priority); 6705 lStatus = effect->addHandle(handle); 6706 if (enabled != NULL) { 6707 *enabled = (int)effect->isEnabled(); 6708 } 6709 } 6710 6711Exit: 6712 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6713 Mutex::Autolock _l(mLock); 6714 if (effectCreated) { 6715 chain->removeEffect_l(effect); 6716 } 6717 if (effectRegistered) { 6718 AudioSystem::unregisterEffect(effect->id()); 6719 } 6720 if (chainCreated) { 6721 removeEffectChain_l(chain); 6722 } 6723 handle.clear(); 6724 } 6725 6726 if (status != NULL) { 6727 *status = lStatus; 6728 } 6729 return handle; 6730} 6731 6732sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6733{ 6734 sp<EffectChain> chain = getEffectChain_l(sessionId); 6735 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6736} 6737 6738// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6739// PlaybackThread::mLock held 6740status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6741{ 6742 // check for existing effect chain with the requested audio session 6743 int sessionId = effect->sessionId(); 6744 sp<EffectChain> chain = getEffectChain_l(sessionId); 6745 bool chainCreated = false; 6746 6747 if (chain == 0) { 6748 // create a new chain for this session 6749 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6750 chain = new EffectChain(this, sessionId); 6751 addEffectChain_l(chain); 6752 chain->setStrategy(getStrategyForSession_l(sessionId)); 6753 chainCreated = true; 6754 } 6755 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6756 6757 if (chain->getEffectFromId_l(effect->id()) != 0) { 6758 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6759 this, effect->desc().name, chain.get()); 6760 return BAD_VALUE; 6761 } 6762 6763 status_t status = chain->addEffect_l(effect); 6764 if (status != NO_ERROR) { 6765 if (chainCreated) { 6766 removeEffectChain_l(chain); 6767 } 6768 return status; 6769 } 6770 6771 effect->setDevice(mDevice); 6772 effect->setMode(mAudioFlinger->getMode()); 6773 return NO_ERROR; 6774} 6775 6776void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6777 6778 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6779 effect_descriptor_t desc = effect->desc(); 6780 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6781 detachAuxEffect_l(effect->id()); 6782 } 6783 6784 sp<EffectChain> chain = effect->chain().promote(); 6785 if (chain != 0) { 6786 // remove effect chain if removing last effect 6787 if (chain->removeEffect_l(effect) == 0) { 6788 removeEffectChain_l(chain); 6789 } 6790 } else { 6791 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6792 } 6793} 6794 6795void AudioFlinger::ThreadBase::lockEffectChains_l( 6796 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6797{ 6798 effectChains = mEffectChains; 6799 for (size_t i = 0; i < mEffectChains.size(); i++) { 6800 mEffectChains[i]->lock(); 6801 } 6802} 6803 6804void AudioFlinger::ThreadBase::unlockEffectChains( 6805 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6806{ 6807 for (size_t i = 0; i < effectChains.size(); i++) { 6808 effectChains[i]->unlock(); 6809 } 6810} 6811 6812sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6813{ 6814 Mutex::Autolock _l(mLock); 6815 return getEffectChain_l(sessionId); 6816} 6817 6818sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6819{ 6820 size_t size = mEffectChains.size(); 6821 for (size_t i = 0; i < size; i++) { 6822 if (mEffectChains[i]->sessionId() == sessionId) { 6823 return mEffectChains[i]; 6824 } 6825 } 6826 return 0; 6827} 6828 6829void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6830{ 6831 Mutex::Autolock _l(mLock); 6832 size_t size = mEffectChains.size(); 6833 for (size_t i = 0; i < size; i++) { 6834 mEffectChains[i]->setMode_l(mode); 6835 } 6836} 6837 6838void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6839 const wp<EffectHandle>& handle, 6840 bool unpinIfLast) { 6841 6842 Mutex::Autolock _l(mLock); 6843 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6844 // delete the effect module if removing last handle on it 6845 if (effect->removeHandle(handle) == 0) { 6846 if (!effect->isPinned() || unpinIfLast) { 6847 removeEffect_l(effect); 6848 AudioSystem::unregisterEffect(effect->id()); 6849 } 6850 } 6851} 6852 6853status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6854{ 6855 int session = chain->sessionId(); 6856 int16_t *buffer = mMixBuffer; 6857 bool ownsBuffer = false; 6858 6859 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6860 if (session > 0) { 6861 // Only one effect chain can be present in direct output thread and it uses 6862 // the mix buffer as input 6863 if (mType != DIRECT) { 6864 size_t numSamples = mFrameCount * mChannelCount; 6865 buffer = new int16_t[numSamples]; 6866 memset(buffer, 0, numSamples * sizeof(int16_t)); 6867 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6868 ownsBuffer = true; 6869 } 6870 6871 // Attach all tracks with same session ID to this chain. 6872 for (size_t i = 0; i < mTracks.size(); ++i) { 6873 sp<Track> track = mTracks[i]; 6874 if (session == track->sessionId()) { 6875 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6876 track->setMainBuffer(buffer); 6877 chain->incTrackCnt(); 6878 } 6879 } 6880 6881 // indicate all active tracks in the chain 6882 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6883 sp<Track> track = mActiveTracks[i].promote(); 6884 if (track == 0) continue; 6885 if (session == track->sessionId()) { 6886 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6887 chain->incActiveTrackCnt(); 6888 } 6889 } 6890 } 6891 6892 chain->setInBuffer(buffer, ownsBuffer); 6893 chain->setOutBuffer(mMixBuffer); 6894 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6895 // chains list in order to be processed last as it contains output stage effects 6896 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6897 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6898 // after track specific effects and before output stage 6899 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6900 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6901 // Effect chain for other sessions are inserted at beginning of effect 6902 // chains list to be processed before output mix effects. Relative order between other 6903 // sessions is not important 6904 size_t size = mEffectChains.size(); 6905 size_t i = 0; 6906 for (i = 0; i < size; i++) { 6907 if (mEffectChains[i]->sessionId() < session) break; 6908 } 6909 mEffectChains.insertAt(chain, i); 6910 checkSuspendOnAddEffectChain_l(chain); 6911 6912 return NO_ERROR; 6913} 6914 6915size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6916{ 6917 int session = chain->sessionId(); 6918 6919 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6920 6921 for (size_t i = 0; i < mEffectChains.size(); i++) { 6922 if (chain == mEffectChains[i]) { 6923 mEffectChains.removeAt(i); 6924 // detach all active tracks from the chain 6925 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6926 sp<Track> track = mActiveTracks[i].promote(); 6927 if (track == 0) continue; 6928 if (session == track->sessionId()) { 6929 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6930 chain.get(), session); 6931 chain->decActiveTrackCnt(); 6932 } 6933 } 6934 6935 // detach all tracks with same session ID from this chain 6936 for (size_t i = 0; i < mTracks.size(); ++i) { 6937 sp<Track> track = mTracks[i]; 6938 if (session == track->sessionId()) { 6939 track->setMainBuffer(mMixBuffer); 6940 chain->decTrackCnt(); 6941 } 6942 } 6943 break; 6944 } 6945 } 6946 return mEffectChains.size(); 6947} 6948 6949status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6950 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6951{ 6952 Mutex::Autolock _l(mLock); 6953 return attachAuxEffect_l(track, EffectId); 6954} 6955 6956status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6957 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6958{ 6959 status_t status = NO_ERROR; 6960 6961 if (EffectId == 0) { 6962 track->setAuxBuffer(0, NULL); 6963 } else { 6964 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6965 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6966 if (effect != 0) { 6967 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6968 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6969 } else { 6970 status = INVALID_OPERATION; 6971 } 6972 } else { 6973 status = BAD_VALUE; 6974 } 6975 } 6976 return status; 6977} 6978 6979void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6980{ 6981 for (size_t i = 0; i < mTracks.size(); ++i) { 6982 sp<Track> track = mTracks[i]; 6983 if (track->auxEffectId() == effectId) { 6984 attachAuxEffect_l(track, 0); 6985 } 6986 } 6987} 6988 6989status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6990{ 6991 // only one chain per input thread 6992 if (mEffectChains.size() != 0) { 6993 return INVALID_OPERATION; 6994 } 6995 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6996 6997 chain->setInBuffer(NULL); 6998 chain->setOutBuffer(NULL); 6999 7000 checkSuspendOnAddEffectChain_l(chain); 7001 7002 mEffectChains.add(chain); 7003 7004 return NO_ERROR; 7005} 7006 7007size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7008{ 7009 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7010 ALOGW_IF(mEffectChains.size() != 1, 7011 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7012 chain.get(), mEffectChains.size(), this); 7013 if (mEffectChains.size() == 1) { 7014 mEffectChains.removeAt(0); 7015 } 7016 return 0; 7017} 7018 7019// ---------------------------------------------------------------------------- 7020// EffectModule implementation 7021// ---------------------------------------------------------------------------- 7022 7023#undef LOG_TAG 7024#define LOG_TAG "AudioFlinger::EffectModule" 7025 7026AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7027 const wp<AudioFlinger::EffectChain>& chain, 7028 effect_descriptor_t *desc, 7029 int id, 7030 int sessionId) 7031 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7032 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7033{ 7034 ALOGV("Constructor %p", this); 7035 int lStatus; 7036 if (thread == NULL) { 7037 return; 7038 } 7039 7040 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7041 7042 // create effect engine from effect factory 7043 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7044 7045 if (mStatus != NO_ERROR) { 7046 return; 7047 } 7048 lStatus = init(); 7049 if (lStatus < 0) { 7050 mStatus = lStatus; 7051 goto Error; 7052 } 7053 7054 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7055 mPinned = true; 7056 } 7057 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7058 return; 7059Error: 7060 EffectRelease(mEffectInterface); 7061 mEffectInterface = NULL; 7062 ALOGV("Constructor Error %d", mStatus); 7063} 7064 7065AudioFlinger::EffectModule::~EffectModule() 7066{ 7067 ALOGV("Destructor %p", this); 7068 if (mEffectInterface != NULL) { 7069 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7070 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7071 sp<ThreadBase> thread = mThread.promote(); 7072 if (thread != 0) { 7073 audio_stream_t *stream = thread->stream(); 7074 if (stream != NULL) { 7075 stream->remove_audio_effect(stream, mEffectInterface); 7076 } 7077 } 7078 } 7079 // release effect engine 7080 EffectRelease(mEffectInterface); 7081 } 7082} 7083 7084status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7085{ 7086 status_t status; 7087 7088 Mutex::Autolock _l(mLock); 7089 int priority = handle->priority(); 7090 size_t size = mHandles.size(); 7091 sp<EffectHandle> h; 7092 size_t i; 7093 for (i = 0; i < size; i++) { 7094 h = mHandles[i].promote(); 7095 if (h == 0) continue; 7096 if (h->priority() <= priority) break; 7097 } 7098 // if inserted in first place, move effect control from previous owner to this handle 7099 if (i == 0) { 7100 bool enabled = false; 7101 if (h != 0) { 7102 enabled = h->enabled(); 7103 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7104 } 7105 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7106 status = NO_ERROR; 7107 } else { 7108 status = ALREADY_EXISTS; 7109 } 7110 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7111 mHandles.insertAt(handle, i); 7112 return status; 7113} 7114 7115size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7116{ 7117 Mutex::Autolock _l(mLock); 7118 size_t size = mHandles.size(); 7119 size_t i; 7120 for (i = 0; i < size; i++) { 7121 if (mHandles[i] == handle) break; 7122 } 7123 if (i == size) { 7124 return size; 7125 } 7126 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7127 7128 bool enabled = false; 7129 EffectHandle *hdl = handle.unsafe_get(); 7130 if (hdl != NULL) { 7131 ALOGV("removeHandle() unsafe_get OK"); 7132 enabled = hdl->enabled(); 7133 } 7134 mHandles.removeAt(i); 7135 size = mHandles.size(); 7136 // if removed from first place, move effect control from this handle to next in line 7137 if (i == 0 && size != 0) { 7138 sp<EffectHandle> h = mHandles[0].promote(); 7139 if (h != 0) { 7140 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7141 } 7142 } 7143 7144 // Prevent calls to process() and other functions on effect interface from now on. 7145 // The effect engine will be released by the destructor when the last strong reference on 7146 // this object is released which can happen after next process is called. 7147 if (size == 0 && !mPinned) { 7148 mState = DESTROYED; 7149 } 7150 7151 return size; 7152} 7153 7154sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7155{ 7156 Mutex::Autolock _l(mLock); 7157 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7158} 7159 7160void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7161{ 7162 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7163 // keep a strong reference on this EffectModule to avoid calling the 7164 // destructor before we exit 7165 sp<EffectModule> keep(this); 7166 { 7167 sp<ThreadBase> thread = mThread.promote(); 7168 if (thread != 0) { 7169 thread->disconnectEffect(keep, handle, unpinIfLast); 7170 } 7171 } 7172} 7173 7174void AudioFlinger::EffectModule::updateState() { 7175 Mutex::Autolock _l(mLock); 7176 7177 switch (mState) { 7178 case RESTART: 7179 reset_l(); 7180 // FALL THROUGH 7181 7182 case STARTING: 7183 // clear auxiliary effect input buffer for next accumulation 7184 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7185 memset(mConfig.inputCfg.buffer.raw, 7186 0, 7187 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7188 } 7189 start_l(); 7190 mState = ACTIVE; 7191 break; 7192 case STOPPING: 7193 stop_l(); 7194 mDisableWaitCnt = mMaxDisableWaitCnt; 7195 mState = STOPPED; 7196 break; 7197 case STOPPED: 7198 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7199 // turn off sequence. 7200 if (--mDisableWaitCnt == 0) { 7201 reset_l(); 7202 mState = IDLE; 7203 } 7204 break; 7205 default: //IDLE , ACTIVE, DESTROYED 7206 break; 7207 } 7208} 7209 7210void AudioFlinger::EffectModule::process() 7211{ 7212 Mutex::Autolock _l(mLock); 7213 7214 if (mState == DESTROYED || mEffectInterface == NULL || 7215 mConfig.inputCfg.buffer.raw == NULL || 7216 mConfig.outputCfg.buffer.raw == NULL) { 7217 return; 7218 } 7219 7220 if (isProcessEnabled()) { 7221 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7222 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7223 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7224 mConfig.inputCfg.buffer.s32, 7225 mConfig.inputCfg.buffer.frameCount/2); 7226 } 7227 7228 // do the actual processing in the effect engine 7229 int ret = (*mEffectInterface)->process(mEffectInterface, 7230 &mConfig.inputCfg.buffer, 7231 &mConfig.outputCfg.buffer); 7232 7233 // force transition to IDLE state when engine is ready 7234 if (mState == STOPPED && ret == -ENODATA) { 7235 mDisableWaitCnt = 1; 7236 } 7237 7238 // clear auxiliary effect input buffer for next accumulation 7239 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7240 memset(mConfig.inputCfg.buffer.raw, 0, 7241 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7242 } 7243 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7244 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7245 // If an insert effect is idle and input buffer is different from output buffer, 7246 // accumulate input onto output 7247 sp<EffectChain> chain = mChain.promote(); 7248 if (chain != 0 && chain->activeTrackCnt() != 0) { 7249 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7250 int16_t *in = mConfig.inputCfg.buffer.s16; 7251 int16_t *out = mConfig.outputCfg.buffer.s16; 7252 for (size_t i = 0; i < frameCnt; i++) { 7253 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7254 } 7255 } 7256 } 7257} 7258 7259void AudioFlinger::EffectModule::reset_l() 7260{ 7261 if (mEffectInterface == NULL) { 7262 return; 7263 } 7264 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7265} 7266 7267status_t AudioFlinger::EffectModule::configure() 7268{ 7269 uint32_t channels; 7270 if (mEffectInterface == NULL) { 7271 return NO_INIT; 7272 } 7273 7274 sp<ThreadBase> thread = mThread.promote(); 7275 if (thread == 0) { 7276 return DEAD_OBJECT; 7277 } 7278 7279 // TODO: handle configuration of effects replacing track process 7280 if (thread->channelCount() == 1) { 7281 channels = AUDIO_CHANNEL_OUT_MONO; 7282 } else { 7283 channels = AUDIO_CHANNEL_OUT_STEREO; 7284 } 7285 7286 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7287 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7288 } else { 7289 mConfig.inputCfg.channels = channels; 7290 } 7291 mConfig.outputCfg.channels = channels; 7292 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7293 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7294 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7295 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7296 mConfig.inputCfg.bufferProvider.cookie = NULL; 7297 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7298 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7299 mConfig.outputCfg.bufferProvider.cookie = NULL; 7300 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7301 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7302 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7303 // Insert effect: 7304 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7305 // always overwrites output buffer: input buffer == output buffer 7306 // - in other sessions: 7307 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7308 // other effect: overwrites output buffer: input buffer == output buffer 7309 // Auxiliary effect: 7310 // accumulates in output buffer: input buffer != output buffer 7311 // Therefore: accumulate <=> input buffer != output buffer 7312 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7313 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7314 } else { 7315 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7316 } 7317 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7318 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7319 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7320 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7321 7322 ALOGV("configure() %p thread %p buffer %p framecount %d", 7323 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7324 7325 status_t cmdStatus; 7326 uint32_t size = sizeof(int); 7327 status_t status = (*mEffectInterface)->command(mEffectInterface, 7328 EFFECT_CMD_SET_CONFIG, 7329 sizeof(effect_config_t), 7330 &mConfig, 7331 &size, 7332 &cmdStatus); 7333 if (status == 0) { 7334 status = cmdStatus; 7335 } 7336 7337 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7338 (1000 * mConfig.outputCfg.buffer.frameCount); 7339 7340 return status; 7341} 7342 7343status_t AudioFlinger::EffectModule::init() 7344{ 7345 Mutex::Autolock _l(mLock); 7346 if (mEffectInterface == NULL) { 7347 return NO_INIT; 7348 } 7349 status_t cmdStatus; 7350 uint32_t size = sizeof(status_t); 7351 status_t status = (*mEffectInterface)->command(mEffectInterface, 7352 EFFECT_CMD_INIT, 7353 0, 7354 NULL, 7355 &size, 7356 &cmdStatus); 7357 if (status == 0) { 7358 status = cmdStatus; 7359 } 7360 return status; 7361} 7362 7363status_t AudioFlinger::EffectModule::start() 7364{ 7365 Mutex::Autolock _l(mLock); 7366 return start_l(); 7367} 7368 7369status_t AudioFlinger::EffectModule::start_l() 7370{ 7371 if (mEffectInterface == NULL) { 7372 return NO_INIT; 7373 } 7374 status_t cmdStatus; 7375 uint32_t size = sizeof(status_t); 7376 status_t status = (*mEffectInterface)->command(mEffectInterface, 7377 EFFECT_CMD_ENABLE, 7378 0, 7379 NULL, 7380 &size, 7381 &cmdStatus); 7382 if (status == 0) { 7383 status = cmdStatus; 7384 } 7385 if (status == 0 && 7386 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7387 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7388 sp<ThreadBase> thread = mThread.promote(); 7389 if (thread != 0) { 7390 audio_stream_t *stream = thread->stream(); 7391 if (stream != NULL) { 7392 stream->add_audio_effect(stream, mEffectInterface); 7393 } 7394 } 7395 } 7396 return status; 7397} 7398 7399status_t AudioFlinger::EffectModule::stop() 7400{ 7401 Mutex::Autolock _l(mLock); 7402 return stop_l(); 7403} 7404 7405status_t AudioFlinger::EffectModule::stop_l() 7406{ 7407 if (mEffectInterface == NULL) { 7408 return NO_INIT; 7409 } 7410 status_t cmdStatus; 7411 uint32_t size = sizeof(status_t); 7412 status_t status = (*mEffectInterface)->command(mEffectInterface, 7413 EFFECT_CMD_DISABLE, 7414 0, 7415 NULL, 7416 &size, 7417 &cmdStatus); 7418 if (status == 0) { 7419 status = cmdStatus; 7420 } 7421 if (status == 0 && 7422 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7423 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7424 sp<ThreadBase> thread = mThread.promote(); 7425 if (thread != 0) { 7426 audio_stream_t *stream = thread->stream(); 7427 if (stream != NULL) { 7428 stream->remove_audio_effect(stream, mEffectInterface); 7429 } 7430 } 7431 } 7432 return status; 7433} 7434 7435status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7436 uint32_t cmdSize, 7437 void *pCmdData, 7438 uint32_t *replySize, 7439 void *pReplyData) 7440{ 7441 Mutex::Autolock _l(mLock); 7442// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7443 7444 if (mState == DESTROYED || mEffectInterface == NULL) { 7445 return NO_INIT; 7446 } 7447 status_t status = (*mEffectInterface)->command(mEffectInterface, 7448 cmdCode, 7449 cmdSize, 7450 pCmdData, 7451 replySize, 7452 pReplyData); 7453 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7454 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7455 for (size_t i = 1; i < mHandles.size(); i++) { 7456 sp<EffectHandle> h = mHandles[i].promote(); 7457 if (h != 0) { 7458 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7459 } 7460 } 7461 } 7462 return status; 7463} 7464 7465status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7466{ 7467 7468 Mutex::Autolock _l(mLock); 7469 ALOGV("setEnabled %p enabled %d", this, enabled); 7470 7471 if (enabled != isEnabled()) { 7472 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7473 if (enabled && status != NO_ERROR) { 7474 return status; 7475 } 7476 7477 switch (mState) { 7478 // going from disabled to enabled 7479 case IDLE: 7480 mState = STARTING; 7481 break; 7482 case STOPPED: 7483 mState = RESTART; 7484 break; 7485 case STOPPING: 7486 mState = ACTIVE; 7487 break; 7488 7489 // going from enabled to disabled 7490 case RESTART: 7491 mState = STOPPED; 7492 break; 7493 case STARTING: 7494 mState = IDLE; 7495 break; 7496 case ACTIVE: 7497 mState = STOPPING; 7498 break; 7499 case DESTROYED: 7500 return NO_ERROR; // simply ignore as we are being destroyed 7501 } 7502 for (size_t i = 1; i < mHandles.size(); i++) { 7503 sp<EffectHandle> h = mHandles[i].promote(); 7504 if (h != 0) { 7505 h->setEnabled(enabled); 7506 } 7507 } 7508 } 7509 return NO_ERROR; 7510} 7511 7512bool AudioFlinger::EffectModule::isEnabled() const 7513{ 7514 switch (mState) { 7515 case RESTART: 7516 case STARTING: 7517 case ACTIVE: 7518 return true; 7519 case IDLE: 7520 case STOPPING: 7521 case STOPPED: 7522 case DESTROYED: 7523 default: 7524 return false; 7525 } 7526} 7527 7528bool AudioFlinger::EffectModule::isProcessEnabled() const 7529{ 7530 switch (mState) { 7531 case RESTART: 7532 case ACTIVE: 7533 case STOPPING: 7534 case STOPPED: 7535 return true; 7536 case IDLE: 7537 case STARTING: 7538 case DESTROYED: 7539 default: 7540 return false; 7541 } 7542} 7543 7544status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7545{ 7546 Mutex::Autolock _l(mLock); 7547 status_t status = NO_ERROR; 7548 7549 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7550 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7551 if (isProcessEnabled() && 7552 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7553 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7554 status_t cmdStatus; 7555 uint32_t volume[2]; 7556 uint32_t *pVolume = NULL; 7557 uint32_t size = sizeof(volume); 7558 volume[0] = *left; 7559 volume[1] = *right; 7560 if (controller) { 7561 pVolume = volume; 7562 } 7563 status = (*mEffectInterface)->command(mEffectInterface, 7564 EFFECT_CMD_SET_VOLUME, 7565 size, 7566 volume, 7567 &size, 7568 pVolume); 7569 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7570 *left = volume[0]; 7571 *right = volume[1]; 7572 } 7573 } 7574 return status; 7575} 7576 7577status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7578{ 7579 Mutex::Autolock _l(mLock); 7580 status_t status = NO_ERROR; 7581 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7582 // audio pre processing modules on RecordThread can receive both output and 7583 // input device indication in the same call 7584 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7585 if (dev) { 7586 status_t cmdStatus; 7587 uint32_t size = sizeof(status_t); 7588 7589 status = (*mEffectInterface)->command(mEffectInterface, 7590 EFFECT_CMD_SET_DEVICE, 7591 sizeof(uint32_t), 7592 &dev, 7593 &size, 7594 &cmdStatus); 7595 if (status == NO_ERROR) { 7596 status = cmdStatus; 7597 } 7598 } 7599 dev = device & AUDIO_DEVICE_IN_ALL; 7600 if (dev) { 7601 status_t cmdStatus; 7602 uint32_t size = sizeof(status_t); 7603 7604 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7605 EFFECT_CMD_SET_INPUT_DEVICE, 7606 sizeof(uint32_t), 7607 &dev, 7608 &size, 7609 &cmdStatus); 7610 if (status2 == NO_ERROR) { 7611 status2 = cmdStatus; 7612 } 7613 if (status == NO_ERROR) { 7614 status = status2; 7615 } 7616 } 7617 } 7618 return status; 7619} 7620 7621status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7622{ 7623 Mutex::Autolock _l(mLock); 7624 status_t status = NO_ERROR; 7625 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7626 status_t cmdStatus; 7627 uint32_t size = sizeof(status_t); 7628 status = (*mEffectInterface)->command(mEffectInterface, 7629 EFFECT_CMD_SET_AUDIO_MODE, 7630 sizeof(audio_mode_t), 7631 &mode, 7632 &size, 7633 &cmdStatus); 7634 if (status == NO_ERROR) { 7635 status = cmdStatus; 7636 } 7637 } 7638 return status; 7639} 7640 7641void AudioFlinger::EffectModule::setSuspended(bool suspended) 7642{ 7643 Mutex::Autolock _l(mLock); 7644 mSuspended = suspended; 7645} 7646 7647bool AudioFlinger::EffectModule::suspended() const 7648{ 7649 Mutex::Autolock _l(mLock); 7650 return mSuspended; 7651} 7652 7653status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7654{ 7655 const size_t SIZE = 256; 7656 char buffer[SIZE]; 7657 String8 result; 7658 7659 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7660 result.append(buffer); 7661 7662 bool locked = tryLock(mLock); 7663 // failed to lock - AudioFlinger is probably deadlocked 7664 if (!locked) { 7665 result.append("\t\tCould not lock Fx mutex:\n"); 7666 } 7667 7668 result.append("\t\tSession Status State Engine:\n"); 7669 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7670 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7671 result.append(buffer); 7672 7673 result.append("\t\tDescriptor:\n"); 7674 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7675 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7676 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7677 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7678 result.append(buffer); 7679 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7680 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7681 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7682 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7683 result.append(buffer); 7684 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7685 mDescriptor.apiVersion, 7686 mDescriptor.flags); 7687 result.append(buffer); 7688 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7689 mDescriptor.name); 7690 result.append(buffer); 7691 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7692 mDescriptor.implementor); 7693 result.append(buffer); 7694 7695 result.append("\t\t- Input configuration:\n"); 7696 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7697 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7698 (uint32_t)mConfig.inputCfg.buffer.raw, 7699 mConfig.inputCfg.buffer.frameCount, 7700 mConfig.inputCfg.samplingRate, 7701 mConfig.inputCfg.channels, 7702 mConfig.inputCfg.format); 7703 result.append(buffer); 7704 7705 result.append("\t\t- Output configuration:\n"); 7706 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7707 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7708 (uint32_t)mConfig.outputCfg.buffer.raw, 7709 mConfig.outputCfg.buffer.frameCount, 7710 mConfig.outputCfg.samplingRate, 7711 mConfig.outputCfg.channels, 7712 mConfig.outputCfg.format); 7713 result.append(buffer); 7714 7715 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7716 result.append(buffer); 7717 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7718 for (size_t i = 0; i < mHandles.size(); ++i) { 7719 sp<EffectHandle> handle = mHandles[i].promote(); 7720 if (handle != 0) { 7721 handle->dump(buffer, SIZE); 7722 result.append(buffer); 7723 } 7724 } 7725 7726 result.append("\n"); 7727 7728 write(fd, result.string(), result.length()); 7729 7730 if (locked) { 7731 mLock.unlock(); 7732 } 7733 7734 return NO_ERROR; 7735} 7736 7737// ---------------------------------------------------------------------------- 7738// EffectHandle implementation 7739// ---------------------------------------------------------------------------- 7740 7741#undef LOG_TAG 7742#define LOG_TAG "AudioFlinger::EffectHandle" 7743 7744AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7745 const sp<AudioFlinger::Client>& client, 7746 const sp<IEffectClient>& effectClient, 7747 int32_t priority) 7748 : BnEffect(), 7749 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7750 mPriority(priority), mHasControl(false), mEnabled(false) 7751{ 7752 ALOGV("constructor %p", this); 7753 7754 if (client == 0) { 7755 return; 7756 } 7757 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7758 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7759 if (mCblkMemory != 0) { 7760 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7761 7762 if (mCblk != NULL) { 7763 new(mCblk) effect_param_cblk_t(); 7764 mBuffer = (uint8_t *)mCblk + bufOffset; 7765 } 7766 } else { 7767 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7768 return; 7769 } 7770} 7771 7772AudioFlinger::EffectHandle::~EffectHandle() 7773{ 7774 ALOGV("Destructor %p", this); 7775 disconnect(false); 7776 ALOGV("Destructor DONE %p", this); 7777} 7778 7779status_t AudioFlinger::EffectHandle::enable() 7780{ 7781 ALOGV("enable %p", this); 7782 if (!mHasControl) return INVALID_OPERATION; 7783 if (mEffect == 0) return DEAD_OBJECT; 7784 7785 if (mEnabled) { 7786 return NO_ERROR; 7787 } 7788 7789 mEnabled = true; 7790 7791 sp<ThreadBase> thread = mEffect->thread().promote(); 7792 if (thread != 0) { 7793 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7794 } 7795 7796 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7797 if (mEffect->suspended()) { 7798 return NO_ERROR; 7799 } 7800 7801 status_t status = mEffect->setEnabled(true); 7802 if (status != NO_ERROR) { 7803 if (thread != 0) { 7804 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7805 } 7806 mEnabled = false; 7807 } 7808 return status; 7809} 7810 7811status_t AudioFlinger::EffectHandle::disable() 7812{ 7813 ALOGV("disable %p", this); 7814 if (!mHasControl) return INVALID_OPERATION; 7815 if (mEffect == 0) return DEAD_OBJECT; 7816 7817 if (!mEnabled) { 7818 return NO_ERROR; 7819 } 7820 mEnabled = false; 7821 7822 if (mEffect->suspended()) { 7823 return NO_ERROR; 7824 } 7825 7826 status_t status = mEffect->setEnabled(false); 7827 7828 sp<ThreadBase> thread = mEffect->thread().promote(); 7829 if (thread != 0) { 7830 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7831 } 7832 7833 return status; 7834} 7835 7836void AudioFlinger::EffectHandle::disconnect() 7837{ 7838 disconnect(true); 7839} 7840 7841void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7842{ 7843 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7844 if (mEffect == 0) { 7845 return; 7846 } 7847 mEffect->disconnect(this, unpinIfLast); 7848 7849 if (mHasControl && mEnabled) { 7850 sp<ThreadBase> thread = mEffect->thread().promote(); 7851 if (thread != 0) { 7852 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7853 } 7854 } 7855 7856 // release sp on module => module destructor can be called now 7857 mEffect.clear(); 7858 if (mClient != 0) { 7859 if (mCblk != NULL) { 7860 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7861 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7862 } 7863 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7864 // Client destructor must run with AudioFlinger mutex locked 7865 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7866 mClient.clear(); 7867 } 7868} 7869 7870status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7871 uint32_t cmdSize, 7872 void *pCmdData, 7873 uint32_t *replySize, 7874 void *pReplyData) 7875{ 7876// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7877// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7878 7879 // only get parameter command is permitted for applications not controlling the effect 7880 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7881 return INVALID_OPERATION; 7882 } 7883 if (mEffect == 0) return DEAD_OBJECT; 7884 if (mClient == 0) return INVALID_OPERATION; 7885 7886 // handle commands that are not forwarded transparently to effect engine 7887 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7888 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7889 // no risk to block the whole media server process or mixer threads is we are stuck here 7890 Mutex::Autolock _l(mCblk->lock); 7891 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7892 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7893 mCblk->serverIndex = 0; 7894 mCblk->clientIndex = 0; 7895 return BAD_VALUE; 7896 } 7897 status_t status = NO_ERROR; 7898 while (mCblk->serverIndex < mCblk->clientIndex) { 7899 int reply; 7900 uint32_t rsize = sizeof(int); 7901 int *p = (int *)(mBuffer + mCblk->serverIndex); 7902 int size = *p++; 7903 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7904 ALOGW("command(): invalid parameter block size"); 7905 break; 7906 } 7907 effect_param_t *param = (effect_param_t *)p; 7908 if (param->psize == 0 || param->vsize == 0) { 7909 ALOGW("command(): null parameter or value size"); 7910 mCblk->serverIndex += size; 7911 continue; 7912 } 7913 uint32_t psize = sizeof(effect_param_t) + 7914 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7915 param->vsize; 7916 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7917 psize, 7918 p, 7919 &rsize, 7920 &reply); 7921 // stop at first error encountered 7922 if (ret != NO_ERROR) { 7923 status = ret; 7924 *(int *)pReplyData = reply; 7925 break; 7926 } else if (reply != NO_ERROR) { 7927 *(int *)pReplyData = reply; 7928 break; 7929 } 7930 mCblk->serverIndex += size; 7931 } 7932 mCblk->serverIndex = 0; 7933 mCblk->clientIndex = 0; 7934 return status; 7935 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7936 *(int *)pReplyData = NO_ERROR; 7937 return enable(); 7938 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7939 *(int *)pReplyData = NO_ERROR; 7940 return disable(); 7941 } 7942 7943 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7944} 7945 7946void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7947{ 7948 ALOGV("setControl %p control %d", this, hasControl); 7949 7950 mHasControl = hasControl; 7951 mEnabled = enabled; 7952 7953 if (signal && mEffectClient != 0) { 7954 mEffectClient->controlStatusChanged(hasControl); 7955 } 7956} 7957 7958void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7959 uint32_t cmdSize, 7960 void *pCmdData, 7961 uint32_t replySize, 7962 void *pReplyData) 7963{ 7964 if (mEffectClient != 0) { 7965 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7966 } 7967} 7968 7969 7970 7971void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7972{ 7973 if (mEffectClient != 0) { 7974 mEffectClient->enableStatusChanged(enabled); 7975 } 7976} 7977 7978status_t AudioFlinger::EffectHandle::onTransact( 7979 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7980{ 7981 return BnEffect::onTransact(code, data, reply, flags); 7982} 7983 7984 7985void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7986{ 7987 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7988 7989 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7990 (mClient == 0) ? getpid_cached : mClient->pid(), 7991 mPriority, 7992 mHasControl, 7993 !locked, 7994 mCblk ? mCblk->clientIndex : 0, 7995 mCblk ? mCblk->serverIndex : 0 7996 ); 7997 7998 if (locked) { 7999 mCblk->lock.unlock(); 8000 } 8001} 8002 8003#undef LOG_TAG 8004#define LOG_TAG "AudioFlinger::EffectChain" 8005 8006AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8007 int sessionId) 8008 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8009 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8010 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8011{ 8012 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8013 if (thread == NULL) { 8014 return; 8015 } 8016 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8017 thread->frameCount(); 8018} 8019 8020AudioFlinger::EffectChain::~EffectChain() 8021{ 8022 if (mOwnInBuffer) { 8023 delete mInBuffer; 8024 } 8025 8026} 8027 8028// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8029sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8030{ 8031 size_t size = mEffects.size(); 8032 8033 for (size_t i = 0; i < size; i++) { 8034 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8035 return mEffects[i]; 8036 } 8037 } 8038 return 0; 8039} 8040 8041// getEffectFromId_l() must be called with ThreadBase::mLock held 8042sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8043{ 8044 size_t size = mEffects.size(); 8045 8046 for (size_t i = 0; i < size; i++) { 8047 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8048 if (id == 0 || mEffects[i]->id() == id) { 8049 return mEffects[i]; 8050 } 8051 } 8052 return 0; 8053} 8054 8055// getEffectFromType_l() must be called with ThreadBase::mLock held 8056sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8057 const effect_uuid_t *type) 8058{ 8059 size_t size = mEffects.size(); 8060 8061 for (size_t i = 0; i < size; i++) { 8062 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8063 return mEffects[i]; 8064 } 8065 } 8066 return 0; 8067} 8068 8069// Must be called with EffectChain::mLock locked 8070void AudioFlinger::EffectChain::process_l() 8071{ 8072 sp<ThreadBase> thread = mThread.promote(); 8073 if (thread == 0) { 8074 ALOGW("process_l(): cannot promote mixer thread"); 8075 return; 8076 } 8077 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8078 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8079 // always process effects unless no more tracks are on the session and the effect tail 8080 // has been rendered 8081 bool doProcess = true; 8082 if (!isGlobalSession) { 8083 bool tracksOnSession = (trackCnt() != 0); 8084 8085 if (!tracksOnSession && mTailBufferCount == 0) { 8086 doProcess = false; 8087 } 8088 8089 if (activeTrackCnt() == 0) { 8090 // if no track is active and the effect tail has not been rendered, 8091 // the input buffer must be cleared here as the mixer process will not do it 8092 if (tracksOnSession || mTailBufferCount > 0) { 8093 size_t numSamples = thread->frameCount() * thread->channelCount(); 8094 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8095 if (mTailBufferCount > 0) { 8096 mTailBufferCount--; 8097 } 8098 } 8099 } 8100 } 8101 8102 size_t size = mEffects.size(); 8103 if (doProcess) { 8104 for (size_t i = 0; i < size; i++) { 8105 mEffects[i]->process(); 8106 } 8107 } 8108 for (size_t i = 0; i < size; i++) { 8109 mEffects[i]->updateState(); 8110 } 8111} 8112 8113// addEffect_l() must be called with PlaybackThread::mLock held 8114status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8115{ 8116 effect_descriptor_t desc = effect->desc(); 8117 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8118 8119 Mutex::Autolock _l(mLock); 8120 effect->setChain(this); 8121 sp<ThreadBase> thread = mThread.promote(); 8122 if (thread == 0) { 8123 return NO_INIT; 8124 } 8125 effect->setThread(thread); 8126 8127 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8128 // Auxiliary effects are inserted at the beginning of mEffects vector as 8129 // they are processed first and accumulated in chain input buffer 8130 mEffects.insertAt(effect, 0); 8131 8132 // the input buffer for auxiliary effect contains mono samples in 8133 // 32 bit format. This is to avoid saturation in AudoMixer 8134 // accumulation stage. Saturation is done in EffectModule::process() before 8135 // calling the process in effect engine 8136 size_t numSamples = thread->frameCount(); 8137 int32_t *buffer = new int32_t[numSamples]; 8138 memset(buffer, 0, numSamples * sizeof(int32_t)); 8139 effect->setInBuffer((int16_t *)buffer); 8140 // auxiliary effects output samples to chain input buffer for further processing 8141 // by insert effects 8142 effect->setOutBuffer(mInBuffer); 8143 } else { 8144 // Insert effects are inserted at the end of mEffects vector as they are processed 8145 // after track and auxiliary effects. 8146 // Insert effect order as a function of indicated preference: 8147 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8148 // another effect is present 8149 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8150 // last effect claiming first position 8151 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8152 // first effect claiming last position 8153 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8154 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8155 // already present 8156 8157 size_t size = mEffects.size(); 8158 size_t idx_insert = size; 8159 ssize_t idx_insert_first = -1; 8160 ssize_t idx_insert_last = -1; 8161 8162 for (size_t i = 0; i < size; i++) { 8163 effect_descriptor_t d = mEffects[i]->desc(); 8164 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8165 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8166 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8167 // check invalid effect chaining combinations 8168 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8169 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8170 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8171 return INVALID_OPERATION; 8172 } 8173 // remember position of first insert effect and by default 8174 // select this as insert position for new effect 8175 if (idx_insert == size) { 8176 idx_insert = i; 8177 } 8178 // remember position of last insert effect claiming 8179 // first position 8180 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8181 idx_insert_first = i; 8182 } 8183 // remember position of first insert effect claiming 8184 // last position 8185 if (iPref == EFFECT_FLAG_INSERT_LAST && 8186 idx_insert_last == -1) { 8187 idx_insert_last = i; 8188 } 8189 } 8190 } 8191 8192 // modify idx_insert from first position if needed 8193 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8194 if (idx_insert_last != -1) { 8195 idx_insert = idx_insert_last; 8196 } else { 8197 idx_insert = size; 8198 } 8199 } else { 8200 if (idx_insert_first != -1) { 8201 idx_insert = idx_insert_first + 1; 8202 } 8203 } 8204 8205 // always read samples from chain input buffer 8206 effect->setInBuffer(mInBuffer); 8207 8208 // if last effect in the chain, output samples to chain 8209 // output buffer, otherwise to chain input buffer 8210 if (idx_insert == size) { 8211 if (idx_insert != 0) { 8212 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8213 mEffects[idx_insert-1]->configure(); 8214 } 8215 effect->setOutBuffer(mOutBuffer); 8216 } else { 8217 effect->setOutBuffer(mInBuffer); 8218 } 8219 mEffects.insertAt(effect, idx_insert); 8220 8221 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8222 } 8223 effect->configure(); 8224 return NO_ERROR; 8225} 8226 8227// removeEffect_l() must be called with PlaybackThread::mLock held 8228size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8229{ 8230 Mutex::Autolock _l(mLock); 8231 size_t size = mEffects.size(); 8232 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8233 8234 for (size_t i = 0; i < size; i++) { 8235 if (effect == mEffects[i]) { 8236 // calling stop here will remove pre-processing effect from the audio HAL. 8237 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8238 // the middle of a read from audio HAL 8239 if (mEffects[i]->state() == EffectModule::ACTIVE || 8240 mEffects[i]->state() == EffectModule::STOPPING) { 8241 mEffects[i]->stop(); 8242 } 8243 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8244 delete[] effect->inBuffer(); 8245 } else { 8246 if (i == size - 1 && i != 0) { 8247 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8248 mEffects[i - 1]->configure(); 8249 } 8250 } 8251 mEffects.removeAt(i); 8252 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8253 break; 8254 } 8255 } 8256 8257 return mEffects.size(); 8258} 8259 8260// setDevice_l() must be called with PlaybackThread::mLock held 8261void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8262{ 8263 size_t size = mEffects.size(); 8264 for (size_t i = 0; i < size; i++) { 8265 mEffects[i]->setDevice(device); 8266 } 8267} 8268 8269// setMode_l() must be called with PlaybackThread::mLock held 8270void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8271{ 8272 size_t size = mEffects.size(); 8273 for (size_t i = 0; i < size; i++) { 8274 mEffects[i]->setMode(mode); 8275 } 8276} 8277 8278// setVolume_l() must be called with PlaybackThread::mLock held 8279bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8280{ 8281 uint32_t newLeft = *left; 8282 uint32_t newRight = *right; 8283 bool hasControl = false; 8284 int ctrlIdx = -1; 8285 size_t size = mEffects.size(); 8286 8287 // first update volume controller 8288 for (size_t i = size; i > 0; i--) { 8289 if (mEffects[i - 1]->isProcessEnabled() && 8290 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8291 ctrlIdx = i - 1; 8292 hasControl = true; 8293 break; 8294 } 8295 } 8296 8297 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8298 if (hasControl) { 8299 *left = mNewLeftVolume; 8300 *right = mNewRightVolume; 8301 } 8302 return hasControl; 8303 } 8304 8305 mVolumeCtrlIdx = ctrlIdx; 8306 mLeftVolume = newLeft; 8307 mRightVolume = newRight; 8308 8309 // second get volume update from volume controller 8310 if (ctrlIdx >= 0) { 8311 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8312 mNewLeftVolume = newLeft; 8313 mNewRightVolume = newRight; 8314 } 8315 // then indicate volume to all other effects in chain. 8316 // Pass altered volume to effects before volume controller 8317 // and requested volume to effects after controller 8318 uint32_t lVol = newLeft; 8319 uint32_t rVol = newRight; 8320 8321 for (size_t i = 0; i < size; i++) { 8322 if ((int)i == ctrlIdx) continue; 8323 // this also works for ctrlIdx == -1 when there is no volume controller 8324 if ((int)i > ctrlIdx) { 8325 lVol = *left; 8326 rVol = *right; 8327 } 8328 mEffects[i]->setVolume(&lVol, &rVol, false); 8329 } 8330 *left = newLeft; 8331 *right = newRight; 8332 8333 return hasControl; 8334} 8335 8336status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8337{ 8338 const size_t SIZE = 256; 8339 char buffer[SIZE]; 8340 String8 result; 8341 8342 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8343 result.append(buffer); 8344 8345 bool locked = tryLock(mLock); 8346 // failed to lock - AudioFlinger is probably deadlocked 8347 if (!locked) { 8348 result.append("\tCould not lock mutex:\n"); 8349 } 8350 8351 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8352 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8353 mEffects.size(), 8354 (uint32_t)mInBuffer, 8355 (uint32_t)mOutBuffer, 8356 mActiveTrackCnt); 8357 result.append(buffer); 8358 write(fd, result.string(), result.size()); 8359 8360 for (size_t i = 0; i < mEffects.size(); ++i) { 8361 sp<EffectModule> effect = mEffects[i]; 8362 if (effect != 0) { 8363 effect->dump(fd, args); 8364 } 8365 } 8366 8367 if (locked) { 8368 mLock.unlock(); 8369 } 8370 8371 return NO_ERROR; 8372} 8373 8374// must be called with ThreadBase::mLock held 8375void AudioFlinger::EffectChain::setEffectSuspended_l( 8376 const effect_uuid_t *type, bool suspend) 8377{ 8378 sp<SuspendedEffectDesc> desc; 8379 // use effect type UUID timelow as key as there is no real risk of identical 8380 // timeLow fields among effect type UUIDs. 8381 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8382 if (suspend) { 8383 if (index >= 0) { 8384 desc = mSuspendedEffects.valueAt(index); 8385 } else { 8386 desc = new SuspendedEffectDesc(); 8387 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8388 mSuspendedEffects.add(type->timeLow, desc); 8389 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8390 } 8391 if (desc->mRefCount++ == 0) { 8392 sp<EffectModule> effect = getEffectIfEnabled(type); 8393 if (effect != 0) { 8394 desc->mEffect = effect; 8395 effect->setSuspended(true); 8396 effect->setEnabled(false); 8397 } 8398 } 8399 } else { 8400 if (index < 0) { 8401 return; 8402 } 8403 desc = mSuspendedEffects.valueAt(index); 8404 if (desc->mRefCount <= 0) { 8405 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8406 desc->mRefCount = 1; 8407 } 8408 if (--desc->mRefCount == 0) { 8409 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8410 if (desc->mEffect != 0) { 8411 sp<EffectModule> effect = desc->mEffect.promote(); 8412 if (effect != 0) { 8413 effect->setSuspended(false); 8414 sp<EffectHandle> handle = effect->controlHandle(); 8415 if (handle != 0) { 8416 effect->setEnabled(handle->enabled()); 8417 } 8418 } 8419 desc->mEffect.clear(); 8420 } 8421 mSuspendedEffects.removeItemsAt(index); 8422 } 8423 } 8424} 8425 8426// must be called with ThreadBase::mLock held 8427void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8428{ 8429 sp<SuspendedEffectDesc> desc; 8430 8431 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8432 if (suspend) { 8433 if (index >= 0) { 8434 desc = mSuspendedEffects.valueAt(index); 8435 } else { 8436 desc = new SuspendedEffectDesc(); 8437 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8438 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8439 } 8440 if (desc->mRefCount++ == 0) { 8441 Vector< sp<EffectModule> > effects; 8442 getSuspendEligibleEffects(effects); 8443 for (size_t i = 0; i < effects.size(); i++) { 8444 setEffectSuspended_l(&effects[i]->desc().type, true); 8445 } 8446 } 8447 } else { 8448 if (index < 0) { 8449 return; 8450 } 8451 desc = mSuspendedEffects.valueAt(index); 8452 if (desc->mRefCount <= 0) { 8453 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8454 desc->mRefCount = 1; 8455 } 8456 if (--desc->mRefCount == 0) { 8457 Vector<const effect_uuid_t *> types; 8458 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8459 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8460 continue; 8461 } 8462 types.add(&mSuspendedEffects.valueAt(i)->mType); 8463 } 8464 for (size_t i = 0; i < types.size(); i++) { 8465 setEffectSuspended_l(types[i], false); 8466 } 8467 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8468 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8469 } 8470 } 8471} 8472 8473 8474// The volume effect is used for automated tests only 8475#ifndef OPENSL_ES_H_ 8476static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8477 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8478const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8479#endif //OPENSL_ES_H_ 8480 8481bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8482{ 8483 // auxiliary effects and visualizer are never suspended on output mix 8484 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8485 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8486 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8487 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8488 return false; 8489 } 8490 return true; 8491} 8492 8493void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8494{ 8495 effects.clear(); 8496 for (size_t i = 0; i < mEffects.size(); i++) { 8497 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8498 effects.add(mEffects[i]); 8499 } 8500 } 8501} 8502 8503sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8504 const effect_uuid_t *type) 8505{ 8506 sp<EffectModule> effect = getEffectFromType_l(type); 8507 return effect != 0 && effect->isEnabled() ? effect : 0; 8508} 8509 8510void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8511 bool enabled) 8512{ 8513 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8514 if (enabled) { 8515 if (index < 0) { 8516 // if the effect is not suspend check if all effects are suspended 8517 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8518 if (index < 0) { 8519 return; 8520 } 8521 if (!isEffectEligibleForSuspend(effect->desc())) { 8522 return; 8523 } 8524 setEffectSuspended_l(&effect->desc().type, enabled); 8525 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8526 if (index < 0) { 8527 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8528 return; 8529 } 8530 } 8531 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8532 effect->desc().type.timeLow); 8533 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8534 // if effect is requested to suspended but was not yet enabled, supend it now. 8535 if (desc->mEffect == 0) { 8536 desc->mEffect = effect; 8537 effect->setEnabled(false); 8538 effect->setSuspended(true); 8539 } 8540 } else { 8541 if (index < 0) { 8542 return; 8543 } 8544 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8545 effect->desc().type.timeLow); 8546 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8547 desc->mEffect.clear(); 8548 effect->setSuspended(false); 8549 } 8550} 8551 8552#undef LOG_TAG 8553#define LOG_TAG "AudioFlinger" 8554 8555// ---------------------------------------------------------------------------- 8556 8557status_t AudioFlinger::onTransact( 8558 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8559{ 8560 return BnAudioFlinger::onTransact(code, data, reply, flags); 8561} 8562 8563}; // namespace android 8564