AudioFlinger.cpp revision 8d314b709fdd81bb64bdaa8d72a0b19c355cefb9
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 mTimedAudioOutputOnTime = false; 4115 return INVALID_OPERATION; 4116 } 4117 4118 Mutex::Autolock _l(mTimedBufferQueueLock); 4119 4120 ALOG_ASSERT(!mQueueHeadInFlight, 4121 "getNextBuffer called without releaseBuffer!"); 4122 4123 while (true) { 4124 4125 // if we have no timed buffers, then fail 4126 if (mTimedBufferQueue.isEmpty()) { 4127 buffer->raw = 0; 4128 buffer->frameCount = 0; 4129 return NOT_ENOUGH_DATA; 4130 } 4131 4132 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4133 4134 // calculate the PTS of the head of the timed buffer queue expressed in 4135 // local time 4136 int64_t headLocalPTS; 4137 { 4138 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4139 4140 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4141 4142 if (mMediaTimeTransform.a_to_b_denom == 0) { 4143 // the transform represents a pause, so yield silence 4144 timedYieldSilence_l(buffer->frameCount, buffer); 4145 return NO_ERROR; 4146 } 4147 4148 int64_t transformedPTS; 4149 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4150 &transformedPTS)) { 4151 // the transform failed. this shouldn't happen, but if it does 4152 // then just drop this buffer 4153 ALOGW("timedGetNextBuffer transform failed"); 4154 buffer->raw = 0; 4155 buffer->frameCount = 0; 4156 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4157 return NO_ERROR; 4158 } 4159 4160 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4161 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4162 &headLocalPTS)) { 4163 buffer->raw = 0; 4164 buffer->frameCount = 0; 4165 return INVALID_OPERATION; 4166 } 4167 } else { 4168 headLocalPTS = transformedPTS; 4169 } 4170 } 4171 4172 // adjust the head buffer's PTS to reflect the portion of the head buffer 4173 // that has already been consumed 4174 int64_t effectivePTS = headLocalPTS + 4175 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4176 4177 // Calculate the delta in samples between the head of the input buffer 4178 // queue and the start of the next output buffer that will be written. 4179 // If the transformation fails because of over or underflow, it means 4180 // that the sample's position in the output stream is so far out of 4181 // whack that it should just be dropped. 4182 int64_t sampleDelta; 4183 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4184 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4185 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4186 " mix"); 4187 continue; 4188 } 4189 if (!mLocalTimeToSampleTransform.doForwardTransform( 4190 (effectivePTS - pts) << 32, &sampleDelta)) { 4191 ALOGV("*** too late during sample rate transform: dropped buffer"); 4192 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 4193 continue; 4194 } 4195 4196 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 4197 " sampleDelta=[%d.%08x]", 4198 head.pts(), head.position(), pts, 4199 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 4200 + (sampleDelta >> 32)), 4201 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4202 4203 // if the delta between the ideal placement for the next input sample and 4204 // the current output position is within this threshold, then we will 4205 // concatenate the next input samples to the previous output 4206 const int64_t kSampleContinuityThreshold = 4207 (static_cast<int64_t>(sampleRate()) << 32) / 250; 4208 4209 // if this is the first buffer of audio that we're emitting from this track 4210 // then it should be almost exactly on time. 4211 const int64_t kSampleStartupThreshold = 1LL << 32; 4212 4213 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4214 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4215 // the next input is close enough to being on time, so concatenate it 4216 // with the last output 4217 timedYieldSamples_l(buffer); 4218 4219 ALOGVV("*** on time: head.pos=%d frameCount=%u", 4220 head.position(), buffer->frameCount); 4221 return NO_ERROR; 4222 } 4223 4224 // Looks like our output is not on time. Reset our on timed status. 4225 // Next time we mix samples from our input queue, then should be within 4226 // the StartupThreshold. 4227 mTimedAudioOutputOnTime = false; 4228 if (sampleDelta > 0) { 4229 // the gap between the current output position and the proper start of 4230 // the next input sample is too big, so fill it with silence 4231 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4232 4233 timedYieldSilence_l(framesUntilNextInput, buffer); 4234 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4235 return NO_ERROR; 4236 } else { 4237 // the next input sample is late 4238 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4239 size_t onTimeSamplePosition = 4240 head.position() + lateFrames * mCblk->frameSize; 4241 4242 if (onTimeSamplePosition > head.buffer()->size()) { 4243 // all the remaining samples in the head are too late, so 4244 // drop it and move on 4245 ALOGV("*** too late: dropped buffer"); 4246 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 4247 continue; 4248 } else { 4249 // skip over the late samples 4250 head.setPosition(onTimeSamplePosition); 4251 4252 // yield the available samples 4253 timedYieldSamples_l(buffer); 4254 4255 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4256 return NO_ERROR; 4257 } 4258 } 4259 } 4260} 4261 4262// Yield samples from the timed buffer queue head up to the given output 4263// buffer's capacity. 4264// 4265// Caller must hold mTimedBufferQueueLock 4266void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 4267 AudioBufferProvider::Buffer* buffer) { 4268 4269 const TimedBuffer& head = mTimedBufferQueue[0]; 4270 4271 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4272 head.position()); 4273 4274 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4275 mCblk->frameSize); 4276 size_t framesRequested = buffer->frameCount; 4277 buffer->frameCount = min(framesLeftInHead, framesRequested); 4278 4279 mQueueHeadInFlight = true; 4280 mTimedAudioOutputOnTime = true; 4281} 4282 4283// Yield samples of silence up to the given output buffer's capacity 4284// 4285// Caller must hold mTimedBufferQueueLock 4286void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 4287 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4288 4289 // lazily allocate a buffer filled with silence 4290 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4291 delete [] mTimedSilenceBuffer; 4292 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4293 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4294 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4295 } 4296 4297 buffer->raw = mTimedSilenceBuffer; 4298 size_t framesRequested = buffer->frameCount; 4299 buffer->frameCount = min(numFrames, framesRequested); 4300 4301 mTimedAudioOutputOnTime = false; 4302} 4303 4304// AudioBufferProvider interface 4305void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4306 AudioBufferProvider::Buffer* buffer) { 4307 4308 Mutex::Autolock _l(mTimedBufferQueueLock); 4309 4310 // If the buffer which was just released is part of the buffer at the head 4311 // of the queue, be sure to update the amt of the buffer which has been 4312 // consumed. If the buffer being returned is not part of the head of the 4313 // queue, its either because the buffer is part of the silence buffer, or 4314 // because the head of the timed queue was trimmed after the mixer called 4315 // getNextBuffer but before the mixer called releaseBuffer. 4316 if (buffer->raw == mTimedSilenceBuffer) { 4317 ALOG_ASSERT(!mQueueHeadInFlight, 4318 "Queue head in flight during release of silence buffer!"); 4319 goto done; 4320 } 4321 4322 ALOG_ASSERT(mQueueHeadInFlight, 4323 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 4324 " head in flight."); 4325 4326 if (mTimedBufferQueue.size()) { 4327 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4328 4329 void* start = head.buffer()->pointer(); 4330 void* end = reinterpret_cast<void*>( 4331 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 4332 + head.buffer()->size()); 4333 4334 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 4335 "released buffer not within the head of the timed buffer" 4336 " queue; qHead = [%p, %p], released buffer = %p", 4337 start, end, buffer->raw); 4338 4339 head.setPosition(head.position() + 4340 (buffer->frameCount * mCblk->frameSize)); 4341 mQueueHeadInFlight = false; 4342 4343 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 4344 "Bad bookkeeping during releaseBuffer! Should have at" 4345 " least %u queued frames, but we think we have only %u", 4346 buffer->frameCount, mFramesPendingInQueue); 4347 4348 mFramesPendingInQueue -= buffer->frameCount; 4349 4350 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 4351 || mTrimQueueHeadOnRelease) { 4352 trimTimedBufferQueueHead_l("releaseBuffer"); 4353 mTrimQueueHeadOnRelease = false; 4354 } 4355 } else { 4356 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 4357 " buffers in the timed buffer queue"); 4358 } 4359 4360done: 4361 buffer->raw = 0; 4362 buffer->frameCount = 0; 4363} 4364 4365uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4366 Mutex::Autolock _l(mTimedBufferQueueLock); 4367 return mFramesPendingInQueue; 4368} 4369 4370AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4371 : mPTS(0), mPosition(0) {} 4372 4373AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4374 const sp<IMemory>& buffer, int64_t pts) 4375 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4376 4377// ---------------------------------------------------------------------------- 4378 4379// RecordTrack constructor must be called with AudioFlinger::mLock held 4380AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4381 RecordThread *thread, 4382 const sp<Client>& client, 4383 uint32_t sampleRate, 4384 audio_format_t format, 4385 uint32_t channelMask, 4386 int frameCount, 4387 int sessionId) 4388 : TrackBase(thread, client, sampleRate, format, 4389 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4390 mOverflow(false) 4391{ 4392 if (mCblk != NULL) { 4393 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4394 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4395 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4396 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4397 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4398 } else { 4399 mCblk->frameSize = sizeof(int8_t); 4400 } 4401 } 4402} 4403 4404AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4405{ 4406 sp<ThreadBase> thread = mThread.promote(); 4407 if (thread != 0) { 4408 AudioSystem::releaseInput(thread->id()); 4409 } 4410} 4411 4412// AudioBufferProvider interface 4413status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4414{ 4415 audio_track_cblk_t* cblk = this->cblk(); 4416 uint32_t framesAvail; 4417 uint32_t framesReq = buffer->frameCount; 4418 4419 // Check if last stepServer failed, try to step now 4420 if (mStepServerFailed) { 4421 if (!step()) goto getNextBuffer_exit; 4422 ALOGV("stepServer recovered"); 4423 mStepServerFailed = false; 4424 } 4425 4426 framesAvail = cblk->framesAvailable_l(); 4427 4428 if (CC_LIKELY(framesAvail)) { 4429 uint32_t s = cblk->server; 4430 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4431 4432 if (framesReq > framesAvail) { 4433 framesReq = framesAvail; 4434 } 4435 if (framesReq > bufferEnd - s) { 4436 framesReq = bufferEnd - s; 4437 } 4438 4439 buffer->raw = getBuffer(s, framesReq); 4440 if (buffer->raw == NULL) goto getNextBuffer_exit; 4441 4442 buffer->frameCount = framesReq; 4443 return NO_ERROR; 4444 } 4445 4446getNextBuffer_exit: 4447 buffer->raw = NULL; 4448 buffer->frameCount = 0; 4449 return NOT_ENOUGH_DATA; 4450} 4451 4452status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid, 4453 AudioSystem::sync_event_t event, 4454 int triggerSession) 4455{ 4456 sp<ThreadBase> thread = mThread.promote(); 4457 if (thread != 0) { 4458 RecordThread *recordThread = (RecordThread *)thread.get(); 4459 return recordThread->start(this, tid, event, triggerSession); 4460 } else { 4461 return BAD_VALUE; 4462 } 4463} 4464 4465void AudioFlinger::RecordThread::RecordTrack::stop() 4466{ 4467 sp<ThreadBase> thread = mThread.promote(); 4468 if (thread != 0) { 4469 RecordThread *recordThread = (RecordThread *)thread.get(); 4470 recordThread->stop(this); 4471 TrackBase::reset(); 4472 // Force overrun condition to avoid false overrun callback until first data is 4473 // read from buffer 4474 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4475 } 4476} 4477 4478void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4479{ 4480 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4481 (mClient == 0) ? getpid_cached : mClient->pid(), 4482 mFormat, 4483 mChannelMask, 4484 mSessionId, 4485 mFrameCount, 4486 mState, 4487 mCblk->sampleRate, 4488 mCblk->server, 4489 mCblk->user); 4490} 4491 4492 4493// ---------------------------------------------------------------------------- 4494 4495AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4496 PlaybackThread *playbackThread, 4497 DuplicatingThread *sourceThread, 4498 uint32_t sampleRate, 4499 audio_format_t format, 4500 uint32_t channelMask, 4501 int frameCount) 4502 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 4503 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 4504 mActive(false), mSourceThread(sourceThread) 4505{ 4506 4507 if (mCblk != NULL) { 4508 mCblk->flags |= CBLK_DIRECTION_OUT; 4509 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4510 mOutBuffer.frameCount = 0; 4511 playbackThread->mTracks.add(this); 4512 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4513 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4514 mCblk, mBuffer, mCblk->buffers, 4515 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4516 } else { 4517 ALOGW("Error creating output track on thread %p", playbackThread); 4518 } 4519} 4520 4521AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4522{ 4523 clearBufferQueue(); 4524} 4525 4526status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid, 4527 AudioSystem::sync_event_t event, 4528 int triggerSession) 4529{ 4530 status_t status = Track::start(tid, event, triggerSession); 4531 if (status != NO_ERROR) { 4532 return status; 4533 } 4534 4535 mActive = true; 4536 mRetryCount = 127; 4537 return status; 4538} 4539 4540void AudioFlinger::PlaybackThread::OutputTrack::stop() 4541{ 4542 Track::stop(); 4543 clearBufferQueue(); 4544 mOutBuffer.frameCount = 0; 4545 mActive = false; 4546} 4547 4548bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4549{ 4550 Buffer *pInBuffer; 4551 Buffer inBuffer; 4552 uint32_t channelCount = mChannelCount; 4553 bool outputBufferFull = false; 4554 inBuffer.frameCount = frames; 4555 inBuffer.i16 = data; 4556 4557 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4558 4559 if (!mActive && frames != 0) { 4560 start(0); 4561 sp<ThreadBase> thread = mThread.promote(); 4562 if (thread != 0) { 4563 MixerThread *mixerThread = (MixerThread *)thread.get(); 4564 if (mCblk->frameCount > frames){ 4565 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4566 uint32_t startFrames = (mCblk->frameCount - frames); 4567 pInBuffer = new Buffer; 4568 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4569 pInBuffer->frameCount = startFrames; 4570 pInBuffer->i16 = pInBuffer->mBuffer; 4571 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4572 mBufferQueue.add(pInBuffer); 4573 } else { 4574 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4575 } 4576 } 4577 } 4578 } 4579 4580 while (waitTimeLeftMs) { 4581 // First write pending buffers, then new data 4582 if (mBufferQueue.size()) { 4583 pInBuffer = mBufferQueue.itemAt(0); 4584 } else { 4585 pInBuffer = &inBuffer; 4586 } 4587 4588 if (pInBuffer->frameCount == 0) { 4589 break; 4590 } 4591 4592 if (mOutBuffer.frameCount == 0) { 4593 mOutBuffer.frameCount = pInBuffer->frameCount; 4594 nsecs_t startTime = systemTime(); 4595 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4596 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4597 outputBufferFull = true; 4598 break; 4599 } 4600 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4601 if (waitTimeLeftMs >= waitTimeMs) { 4602 waitTimeLeftMs -= waitTimeMs; 4603 } else { 4604 waitTimeLeftMs = 0; 4605 } 4606 } 4607 4608 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4609 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4610 mCblk->stepUser(outFrames); 4611 pInBuffer->frameCount -= outFrames; 4612 pInBuffer->i16 += outFrames * channelCount; 4613 mOutBuffer.frameCount -= outFrames; 4614 mOutBuffer.i16 += outFrames * channelCount; 4615 4616 if (pInBuffer->frameCount == 0) { 4617 if (mBufferQueue.size()) { 4618 mBufferQueue.removeAt(0); 4619 delete [] pInBuffer->mBuffer; 4620 delete pInBuffer; 4621 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4622 } else { 4623 break; 4624 } 4625 } 4626 } 4627 4628 // If we could not write all frames, allocate a buffer and queue it for next time. 4629 if (inBuffer.frameCount) { 4630 sp<ThreadBase> thread = mThread.promote(); 4631 if (thread != 0 && !thread->standby()) { 4632 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4633 pInBuffer = new Buffer; 4634 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4635 pInBuffer->frameCount = inBuffer.frameCount; 4636 pInBuffer->i16 = pInBuffer->mBuffer; 4637 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4638 mBufferQueue.add(pInBuffer); 4639 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4640 } else { 4641 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4642 } 4643 } 4644 } 4645 4646 // Calling write() with a 0 length buffer, means that no more data will be written: 4647 // If no more buffers are pending, fill output track buffer to make sure it is started 4648 // by output mixer. 4649 if (frames == 0 && mBufferQueue.size() == 0) { 4650 if (mCblk->user < mCblk->frameCount) { 4651 frames = mCblk->frameCount - mCblk->user; 4652 pInBuffer = new Buffer; 4653 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4654 pInBuffer->frameCount = frames; 4655 pInBuffer->i16 = pInBuffer->mBuffer; 4656 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4657 mBufferQueue.add(pInBuffer); 4658 } else if (mActive) { 4659 stop(); 4660 } 4661 } 4662 4663 return outputBufferFull; 4664} 4665 4666status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4667{ 4668 int active; 4669 status_t result; 4670 audio_track_cblk_t* cblk = mCblk; 4671 uint32_t framesReq = buffer->frameCount; 4672 4673// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4674 buffer->frameCount = 0; 4675 4676 uint32_t framesAvail = cblk->framesAvailable(); 4677 4678 4679 if (framesAvail == 0) { 4680 Mutex::Autolock _l(cblk->lock); 4681 goto start_loop_here; 4682 while (framesAvail == 0) { 4683 active = mActive; 4684 if (CC_UNLIKELY(!active)) { 4685 ALOGV("Not active and NO_MORE_BUFFERS"); 4686 return NO_MORE_BUFFERS; 4687 } 4688 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4689 if (result != NO_ERROR) { 4690 return NO_MORE_BUFFERS; 4691 } 4692 // read the server count again 4693 start_loop_here: 4694 framesAvail = cblk->framesAvailable_l(); 4695 } 4696 } 4697 4698// if (framesAvail < framesReq) { 4699// return NO_MORE_BUFFERS; 4700// } 4701 4702 if (framesReq > framesAvail) { 4703 framesReq = framesAvail; 4704 } 4705 4706 uint32_t u = cblk->user; 4707 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4708 4709 if (framesReq > bufferEnd - u) { 4710 framesReq = bufferEnd - u; 4711 } 4712 4713 buffer->frameCount = framesReq; 4714 buffer->raw = (void *)cblk->buffer(u); 4715 return NO_ERROR; 4716} 4717 4718 4719void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4720{ 4721 size_t size = mBufferQueue.size(); 4722 4723 for (size_t i = 0; i < size; i++) { 4724 Buffer *pBuffer = mBufferQueue.itemAt(i); 4725 delete [] pBuffer->mBuffer; 4726 delete pBuffer; 4727 } 4728 mBufferQueue.clear(); 4729} 4730 4731// ---------------------------------------------------------------------------- 4732 4733AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4734 : RefBase(), 4735 mAudioFlinger(audioFlinger), 4736 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4737 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4738 mPid(pid), 4739 mTimedTrackCount(0) 4740{ 4741 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4742} 4743 4744// Client destructor must be called with AudioFlinger::mLock held 4745AudioFlinger::Client::~Client() 4746{ 4747 mAudioFlinger->removeClient_l(mPid); 4748} 4749 4750sp<MemoryDealer> AudioFlinger::Client::heap() const 4751{ 4752 return mMemoryDealer; 4753} 4754 4755// Reserve one of the limited slots for a timed audio track associated 4756// with this client 4757bool AudioFlinger::Client::reserveTimedTrack() 4758{ 4759 const int kMaxTimedTracksPerClient = 4; 4760 4761 Mutex::Autolock _l(mTimedTrackLock); 4762 4763 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4764 ALOGW("can not create timed track - pid %d has exceeded the limit", 4765 mPid); 4766 return false; 4767 } 4768 4769 mTimedTrackCount++; 4770 return true; 4771} 4772 4773// Release a slot for a timed audio track 4774void AudioFlinger::Client::releaseTimedTrack() 4775{ 4776 Mutex::Autolock _l(mTimedTrackLock); 4777 mTimedTrackCount--; 4778} 4779 4780// ---------------------------------------------------------------------------- 4781 4782AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4783 const sp<IAudioFlingerClient>& client, 4784 pid_t pid) 4785 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4786{ 4787} 4788 4789AudioFlinger::NotificationClient::~NotificationClient() 4790{ 4791} 4792 4793void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4794{ 4795 sp<NotificationClient> keep(this); 4796 mAudioFlinger->removeNotificationClient(mPid); 4797} 4798 4799// ---------------------------------------------------------------------------- 4800 4801AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4802 : BnAudioTrack(), 4803 mTrack(track) 4804{ 4805} 4806 4807AudioFlinger::TrackHandle::~TrackHandle() { 4808 // just stop the track on deletion, associated resources 4809 // will be freed from the main thread once all pending buffers have 4810 // been played. Unless it's not in the active track list, in which 4811 // case we free everything now... 4812 mTrack->destroy(); 4813} 4814 4815sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4816 return mTrack->getCblk(); 4817} 4818 4819status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4820 return mTrack->start(tid); 4821} 4822 4823void AudioFlinger::TrackHandle::stop() { 4824 mTrack->stop(); 4825} 4826 4827void AudioFlinger::TrackHandle::flush() { 4828 mTrack->flush(); 4829} 4830 4831void AudioFlinger::TrackHandle::mute(bool e) { 4832 mTrack->mute(e); 4833} 4834 4835void AudioFlinger::TrackHandle::pause() { 4836 mTrack->pause(); 4837} 4838 4839status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4840{ 4841 return mTrack->attachAuxEffect(EffectId); 4842} 4843 4844status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4845 sp<IMemory>* buffer) { 4846 if (!mTrack->isTimedTrack()) 4847 return INVALID_OPERATION; 4848 4849 PlaybackThread::TimedTrack* tt = 4850 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4851 return tt->allocateTimedBuffer(size, buffer); 4852} 4853 4854status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4855 int64_t pts) { 4856 if (!mTrack->isTimedTrack()) 4857 return INVALID_OPERATION; 4858 4859 PlaybackThread::TimedTrack* tt = 4860 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4861 return tt->queueTimedBuffer(buffer, pts); 4862} 4863 4864status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4865 const LinearTransform& xform, int target) { 4866 4867 if (!mTrack->isTimedTrack()) 4868 return INVALID_OPERATION; 4869 4870 PlaybackThread::TimedTrack* tt = 4871 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4872 return tt->setMediaTimeTransform( 4873 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4874} 4875 4876status_t AudioFlinger::TrackHandle::onTransact( 4877 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4878{ 4879 return BnAudioTrack::onTransact(code, data, reply, flags); 4880} 4881 4882// ---------------------------------------------------------------------------- 4883 4884sp<IAudioRecord> AudioFlinger::openRecord( 4885 pid_t pid, 4886 audio_io_handle_t input, 4887 uint32_t sampleRate, 4888 audio_format_t format, 4889 uint32_t channelMask, 4890 int frameCount, 4891 IAudioFlinger::track_flags_t flags, 4892 int *sessionId, 4893 status_t *status) 4894{ 4895 sp<RecordThread::RecordTrack> recordTrack; 4896 sp<RecordHandle> recordHandle; 4897 sp<Client> client; 4898 status_t lStatus; 4899 RecordThread *thread; 4900 size_t inFrameCount; 4901 int lSessionId; 4902 4903 // check calling permissions 4904 if (!recordingAllowed()) { 4905 lStatus = PERMISSION_DENIED; 4906 goto Exit; 4907 } 4908 4909 // add client to list 4910 { // scope for mLock 4911 Mutex::Autolock _l(mLock); 4912 thread = checkRecordThread_l(input); 4913 if (thread == NULL) { 4914 lStatus = BAD_VALUE; 4915 goto Exit; 4916 } 4917 4918 client = registerPid_l(pid); 4919 4920 // If no audio session id is provided, create one here 4921 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4922 lSessionId = *sessionId; 4923 } else { 4924 lSessionId = nextUniqueId(); 4925 if (sessionId != NULL) { 4926 *sessionId = lSessionId; 4927 } 4928 } 4929 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4930 recordTrack = thread->createRecordTrack_l(client, 4931 sampleRate, 4932 format, 4933 channelMask, 4934 frameCount, 4935 lSessionId, 4936 &lStatus); 4937 } 4938 if (lStatus != NO_ERROR) { 4939 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4940 // destructor is called by the TrackBase destructor with mLock held 4941 client.clear(); 4942 recordTrack.clear(); 4943 goto Exit; 4944 } 4945 4946 // return to handle to client 4947 recordHandle = new RecordHandle(recordTrack); 4948 lStatus = NO_ERROR; 4949 4950Exit: 4951 if (status) { 4952 *status = lStatus; 4953 } 4954 return recordHandle; 4955} 4956 4957// ---------------------------------------------------------------------------- 4958 4959AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4960 : BnAudioRecord(), 4961 mRecordTrack(recordTrack) 4962{ 4963} 4964 4965AudioFlinger::RecordHandle::~RecordHandle() { 4966 stop(); 4967} 4968 4969sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4970 return mRecordTrack->getCblk(); 4971} 4972 4973status_t AudioFlinger::RecordHandle::start(pid_t tid, int event, int triggerSession) { 4974 ALOGV("RecordHandle::start()"); 4975 return mRecordTrack->start(tid, (AudioSystem::sync_event_t)event, triggerSession); 4976} 4977 4978void AudioFlinger::RecordHandle::stop() { 4979 ALOGV("RecordHandle::stop()"); 4980 mRecordTrack->stop(); 4981} 4982 4983status_t AudioFlinger::RecordHandle::onTransact( 4984 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4985{ 4986 return BnAudioRecord::onTransact(code, data, reply, flags); 4987} 4988 4989// ---------------------------------------------------------------------------- 4990 4991AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4992 AudioStreamIn *input, 4993 uint32_t sampleRate, 4994 uint32_t channels, 4995 audio_io_handle_t id, 4996 uint32_t device) : 4997 ThreadBase(audioFlinger, id, device, RECORD), 4998 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4999 // mRsmpInIndex and mInputBytes set by readInputParameters() 5000 mReqChannelCount(popcount(channels)), 5001 mReqSampleRate(sampleRate) 5002 // mBytesRead is only meaningful while active, and so is cleared in start() 5003 // (but might be better to also clear here for dump?) 5004{ 5005 snprintf(mName, kNameLength, "AudioIn_%X", id); 5006 5007 readInputParameters(); 5008} 5009 5010 5011AudioFlinger::RecordThread::~RecordThread() 5012{ 5013 delete[] mRsmpInBuffer; 5014 delete mResampler; 5015 delete[] mRsmpOutBuffer; 5016} 5017 5018void AudioFlinger::RecordThread::onFirstRef() 5019{ 5020 run(mName, PRIORITY_URGENT_AUDIO); 5021} 5022 5023status_t AudioFlinger::RecordThread::readyToRun() 5024{ 5025 status_t status = initCheck(); 5026 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5027 return status; 5028} 5029 5030bool AudioFlinger::RecordThread::threadLoop() 5031{ 5032 AudioBufferProvider::Buffer buffer; 5033 sp<RecordTrack> activeTrack; 5034 Vector< sp<EffectChain> > effectChains; 5035 5036 nsecs_t lastWarning = 0; 5037 5038 acquireWakeLock(); 5039 5040 // start recording 5041 while (!exitPending()) { 5042 5043 processConfigEvents(); 5044 5045 { // scope for mLock 5046 Mutex::Autolock _l(mLock); 5047 checkForNewParameters_l(); 5048 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5049 if (!mStandby) { 5050 mInput->stream->common.standby(&mInput->stream->common); 5051 mStandby = true; 5052 } 5053 5054 if (exitPending()) break; 5055 5056 releaseWakeLock_l(); 5057 ALOGV("RecordThread: loop stopping"); 5058 // go to sleep 5059 mWaitWorkCV.wait(mLock); 5060 ALOGV("RecordThread: loop starting"); 5061 acquireWakeLock_l(); 5062 continue; 5063 } 5064 if (mActiveTrack != 0) { 5065 if (mActiveTrack->mState == TrackBase::PAUSING) { 5066 if (!mStandby) { 5067 mInput->stream->common.standby(&mInput->stream->common); 5068 mStandby = true; 5069 } 5070 mActiveTrack.clear(); 5071 mStartStopCond.broadcast(); 5072 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5073 if (mReqChannelCount != mActiveTrack->channelCount()) { 5074 mActiveTrack.clear(); 5075 mStartStopCond.broadcast(); 5076 } else if (mBytesRead != 0) { 5077 // record start succeeds only if first read from audio input 5078 // succeeds 5079 if (mBytesRead > 0) { 5080 mActiveTrack->mState = TrackBase::ACTIVE; 5081 } else { 5082 mActiveTrack.clear(); 5083 } 5084 mStartStopCond.broadcast(); 5085 } 5086 mStandby = false; 5087 } 5088 } 5089 lockEffectChains_l(effectChains); 5090 } 5091 5092 if (mActiveTrack != 0) { 5093 if (mActiveTrack->mState != TrackBase::ACTIVE && 5094 mActiveTrack->mState != TrackBase::RESUMING) { 5095 unlockEffectChains(effectChains); 5096 usleep(kRecordThreadSleepUs); 5097 continue; 5098 } 5099 for (size_t i = 0; i < effectChains.size(); i ++) { 5100 effectChains[i]->process_l(); 5101 } 5102 5103 buffer.frameCount = mFrameCount; 5104 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5105 size_t framesOut = buffer.frameCount; 5106 if (mResampler == NULL) { 5107 // no resampling 5108 while (framesOut) { 5109 size_t framesIn = mFrameCount - mRsmpInIndex; 5110 if (framesIn) { 5111 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5112 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5113 if (framesIn > framesOut) 5114 framesIn = framesOut; 5115 mRsmpInIndex += framesIn; 5116 framesOut -= framesIn; 5117 if ((int)mChannelCount == mReqChannelCount || 5118 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5119 memcpy(dst, src, framesIn * mFrameSize); 5120 } else { 5121 int16_t *src16 = (int16_t *)src; 5122 int16_t *dst16 = (int16_t *)dst; 5123 if (mChannelCount == 1) { 5124 while (framesIn--) { 5125 *dst16++ = *src16; 5126 *dst16++ = *src16++; 5127 } 5128 } else { 5129 while (framesIn--) { 5130 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5131 src16 += 2; 5132 } 5133 } 5134 } 5135 } 5136 if (framesOut && mFrameCount == mRsmpInIndex) { 5137 if (framesOut == mFrameCount && 5138 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5139 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5140 framesOut = 0; 5141 } else { 5142 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5143 mRsmpInIndex = 0; 5144 } 5145 if (mBytesRead < 0) { 5146 ALOGE("Error reading audio input"); 5147 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5148 // Force input into standby so that it tries to 5149 // recover at next read attempt 5150 mInput->stream->common.standby(&mInput->stream->common); 5151 usleep(kRecordThreadSleepUs); 5152 } 5153 mRsmpInIndex = mFrameCount; 5154 framesOut = 0; 5155 buffer.frameCount = 0; 5156 } 5157 } 5158 } 5159 } else { 5160 // resampling 5161 5162 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5163 // alter output frame count as if we were expecting stereo samples 5164 if (mChannelCount == 1 && mReqChannelCount == 1) { 5165 framesOut >>= 1; 5166 } 5167 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5168 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5169 // are 32 bit aligned which should be always true. 5170 if (mChannelCount == 2 && mReqChannelCount == 1) { 5171 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5172 // the resampler always outputs stereo samples: do post stereo to mono conversion 5173 int16_t *src = (int16_t *)mRsmpOutBuffer; 5174 int16_t *dst = buffer.i16; 5175 while (framesOut--) { 5176 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5177 src += 2; 5178 } 5179 } else { 5180 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5181 } 5182 5183 } 5184 if (mFramestoDrop == 0) { 5185 mActiveTrack->releaseBuffer(&buffer); 5186 } else { 5187 if (mFramestoDrop > 0) { 5188 mFramestoDrop -= buffer.frameCount; 5189 if (mFramestoDrop < 0) { 5190 mFramestoDrop = 0; 5191 } 5192 } 5193 } 5194 mActiveTrack->overflow(); 5195 } 5196 // client isn't retrieving buffers fast enough 5197 else { 5198 if (!mActiveTrack->setOverflow()) { 5199 nsecs_t now = systemTime(); 5200 if ((now - lastWarning) > kWarningThrottleNs) { 5201 ALOGW("RecordThread: buffer overflow"); 5202 lastWarning = now; 5203 } 5204 } 5205 // Release the processor for a while before asking for a new buffer. 5206 // This will give the application more chance to read from the buffer and 5207 // clear the overflow. 5208 usleep(kRecordThreadSleepUs); 5209 } 5210 } 5211 // enable changes in effect chain 5212 unlockEffectChains(effectChains); 5213 effectChains.clear(); 5214 } 5215 5216 if (!mStandby) { 5217 mInput->stream->common.standby(&mInput->stream->common); 5218 } 5219 mActiveTrack.clear(); 5220 5221 mStartStopCond.broadcast(); 5222 5223 releaseWakeLock(); 5224 5225 ALOGV("RecordThread %p exiting", this); 5226 return false; 5227} 5228 5229 5230sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5231 const sp<AudioFlinger::Client>& client, 5232 uint32_t sampleRate, 5233 audio_format_t format, 5234 int channelMask, 5235 int frameCount, 5236 int sessionId, 5237 status_t *status) 5238{ 5239 sp<RecordTrack> track; 5240 status_t lStatus; 5241 5242 lStatus = initCheck(); 5243 if (lStatus != NO_ERROR) { 5244 ALOGE("Audio driver not initialized."); 5245 goto Exit; 5246 } 5247 5248 { // scope for mLock 5249 Mutex::Autolock _l(mLock); 5250 5251 track = new RecordTrack(this, client, sampleRate, 5252 format, channelMask, frameCount, sessionId); 5253 5254 if (track->getCblk() == 0) { 5255 lStatus = NO_MEMORY; 5256 goto Exit; 5257 } 5258 5259 mTrack = track.get(); 5260 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5261 bool suspend = audio_is_bluetooth_sco_device( 5262 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5263 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5264 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5265 } 5266 lStatus = NO_ERROR; 5267 5268Exit: 5269 if (status) { 5270 *status = lStatus; 5271 } 5272 return track; 5273} 5274 5275status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5276 pid_t tid, AudioSystem::sync_event_t event, 5277 int triggerSession) 5278{ 5279 ALOGV("RecordThread::start tid=%d, event %d, triggerSession %d", tid, event, triggerSession); 5280 sp<ThreadBase> strongMe = this; 5281 status_t status = NO_ERROR; 5282 5283 if (event == AudioSystem::SYNC_EVENT_NONE) { 5284 mSyncStartEvent.clear(); 5285 mFramestoDrop = 0; 5286 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5287 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5288 triggerSession, 5289 recordTrack->sessionId(), 5290 syncStartEventCallback, 5291 this); 5292 mFramestoDrop = -1; 5293 } 5294 5295 { 5296 AutoMutex lock(mLock); 5297 if (mActiveTrack != 0) { 5298 if (recordTrack != mActiveTrack.get()) { 5299 status = -EBUSY; 5300 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5301 mActiveTrack->mState = TrackBase::ACTIVE; 5302 } 5303 return status; 5304 } 5305 5306 recordTrack->mState = TrackBase::IDLE; 5307 mActiveTrack = recordTrack; 5308 mLock.unlock(); 5309 status_t status = AudioSystem::startInput(mId); 5310 mLock.lock(); 5311 if (status != NO_ERROR) { 5312 mActiveTrack.clear(); 5313 clearSyncStartEvent(); 5314 return status; 5315 } 5316 mRsmpInIndex = mFrameCount; 5317 mBytesRead = 0; 5318 if (mResampler != NULL) { 5319 mResampler->reset(); 5320 } 5321 mActiveTrack->mState = TrackBase::RESUMING; 5322 // signal thread to start 5323 ALOGV("Signal record thread"); 5324 mWaitWorkCV.signal(); 5325 // do not wait for mStartStopCond if exiting 5326 if (exitPending()) { 5327 mActiveTrack.clear(); 5328 status = INVALID_OPERATION; 5329 goto startError; 5330 } 5331 mStartStopCond.wait(mLock); 5332 if (mActiveTrack == 0) { 5333 ALOGV("Record failed to start"); 5334 status = BAD_VALUE; 5335 goto startError; 5336 } 5337 ALOGV("Record started OK"); 5338 return status; 5339 } 5340startError: 5341 AudioSystem::stopInput(mId); 5342 clearSyncStartEvent(); 5343 return status; 5344} 5345 5346void AudioFlinger::RecordThread::clearSyncStartEvent() 5347{ 5348 if (mSyncStartEvent != 0) { 5349 mSyncStartEvent->cancel(); 5350 } 5351 mSyncStartEvent.clear(); 5352} 5353 5354void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5355{ 5356 sp<SyncEvent> strongEvent = event.promote(); 5357 5358 if (strongEvent != 0) { 5359 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5360 me->handleSyncStartEvent(strongEvent); 5361 } 5362} 5363 5364void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5365{ 5366 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5367 mActiveTrack.get(), 5368 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5369 event->listenerSession()); 5370 5371 if (mActiveTrack != 0 && 5372 event == mSyncStartEvent) { 5373 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5374 // from audio HAL 5375 mFramestoDrop = mFrameCount * 2; 5376 mSyncStartEvent.clear(); 5377 } 5378} 5379 5380void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5381 ALOGV("RecordThread::stop"); 5382 sp<ThreadBase> strongMe = this; 5383 { 5384 AutoMutex lock(mLock); 5385 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5386 mActiveTrack->mState = TrackBase::PAUSING; 5387 // do not wait for mStartStopCond if exiting 5388 if (exitPending()) { 5389 return; 5390 } 5391 mStartStopCond.wait(mLock); 5392 // if we have been restarted, recordTrack == mActiveTrack.get() here 5393 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5394 mLock.unlock(); 5395 AudioSystem::stopInput(mId); 5396 mLock.lock(); 5397 ALOGV("Record stopped OK"); 5398 } 5399 } 5400 } 5401} 5402 5403bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5404{ 5405 return false; 5406} 5407 5408status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5409{ 5410 if (!isValidSyncEvent(event)) { 5411 return BAD_VALUE; 5412 } 5413 5414 Mutex::Autolock _l(mLock); 5415 5416 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5417 mTrack->setSyncEvent(event); 5418 return NO_ERROR; 5419 } 5420 return NAME_NOT_FOUND; 5421} 5422 5423status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5424{ 5425 const size_t SIZE = 256; 5426 char buffer[SIZE]; 5427 String8 result; 5428 5429 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5430 result.append(buffer); 5431 5432 if (mActiveTrack != 0) { 5433 result.append("Active Track:\n"); 5434 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5435 mActiveTrack->dump(buffer, SIZE); 5436 result.append(buffer); 5437 5438 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5439 result.append(buffer); 5440 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5441 result.append(buffer); 5442 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5443 result.append(buffer); 5444 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5445 result.append(buffer); 5446 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5447 result.append(buffer); 5448 5449 5450 } else { 5451 result.append("No record client\n"); 5452 } 5453 write(fd, result.string(), result.size()); 5454 5455 dumpBase(fd, args); 5456 dumpEffectChains(fd, args); 5457 5458 return NO_ERROR; 5459} 5460 5461// AudioBufferProvider interface 5462status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5463{ 5464 size_t framesReq = buffer->frameCount; 5465 size_t framesReady = mFrameCount - mRsmpInIndex; 5466 int channelCount; 5467 5468 if (framesReady == 0) { 5469 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5470 if (mBytesRead < 0) { 5471 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5472 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5473 // Force input into standby so that it tries to 5474 // recover at next read attempt 5475 mInput->stream->common.standby(&mInput->stream->common); 5476 usleep(kRecordThreadSleepUs); 5477 } 5478 buffer->raw = NULL; 5479 buffer->frameCount = 0; 5480 return NOT_ENOUGH_DATA; 5481 } 5482 mRsmpInIndex = 0; 5483 framesReady = mFrameCount; 5484 } 5485 5486 if (framesReq > framesReady) { 5487 framesReq = framesReady; 5488 } 5489 5490 if (mChannelCount == 1 && mReqChannelCount == 2) { 5491 channelCount = 1; 5492 } else { 5493 channelCount = 2; 5494 } 5495 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5496 buffer->frameCount = framesReq; 5497 return NO_ERROR; 5498} 5499 5500// AudioBufferProvider interface 5501void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5502{ 5503 mRsmpInIndex += buffer->frameCount; 5504 buffer->frameCount = 0; 5505} 5506 5507bool AudioFlinger::RecordThread::checkForNewParameters_l() 5508{ 5509 bool reconfig = false; 5510 5511 while (!mNewParameters.isEmpty()) { 5512 status_t status = NO_ERROR; 5513 String8 keyValuePair = mNewParameters[0]; 5514 AudioParameter param = AudioParameter(keyValuePair); 5515 int value; 5516 audio_format_t reqFormat = mFormat; 5517 int reqSamplingRate = mReqSampleRate; 5518 int reqChannelCount = mReqChannelCount; 5519 5520 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5521 reqSamplingRate = value; 5522 reconfig = true; 5523 } 5524 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5525 reqFormat = (audio_format_t) value; 5526 reconfig = true; 5527 } 5528 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5529 reqChannelCount = popcount(value); 5530 reconfig = true; 5531 } 5532 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5533 // do not accept frame count changes if tracks are open as the track buffer 5534 // size depends on frame count and correct behavior would not be guaranteed 5535 // if frame count is changed after track creation 5536 if (mActiveTrack != 0) { 5537 status = INVALID_OPERATION; 5538 } else { 5539 reconfig = true; 5540 } 5541 } 5542 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5543 // forward device change to effects that have requested to be 5544 // aware of attached audio device. 5545 for (size_t i = 0; i < mEffectChains.size(); i++) { 5546 mEffectChains[i]->setDevice_l(value); 5547 } 5548 // store input device and output device but do not forward output device to audio HAL. 5549 // Note that status is ignored by the caller for output device 5550 // (see AudioFlinger::setParameters() 5551 if (value & AUDIO_DEVICE_OUT_ALL) { 5552 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5553 status = BAD_VALUE; 5554 } else { 5555 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5556 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5557 if (mTrack != NULL) { 5558 bool suspend = audio_is_bluetooth_sco_device( 5559 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5560 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5561 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5562 } 5563 } 5564 mDevice |= (uint32_t)value; 5565 } 5566 if (status == NO_ERROR) { 5567 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5568 if (status == INVALID_OPERATION) { 5569 mInput->stream->common.standby(&mInput->stream->common); 5570 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5571 keyValuePair.string()); 5572 } 5573 if (reconfig) { 5574 if (status == BAD_VALUE && 5575 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5576 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5577 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5578 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5579 (reqChannelCount <= FCC_2)) { 5580 status = NO_ERROR; 5581 } 5582 if (status == NO_ERROR) { 5583 readInputParameters(); 5584 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5585 } 5586 } 5587 } 5588 5589 mNewParameters.removeAt(0); 5590 5591 mParamStatus = status; 5592 mParamCond.signal(); 5593 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5594 // already timed out waiting for the status and will never signal the condition. 5595 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5596 } 5597 return reconfig; 5598} 5599 5600String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5601{ 5602 char *s; 5603 String8 out_s8 = String8(); 5604 5605 Mutex::Autolock _l(mLock); 5606 if (initCheck() != NO_ERROR) { 5607 return out_s8; 5608 } 5609 5610 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5611 out_s8 = String8(s); 5612 free(s); 5613 return out_s8; 5614} 5615 5616void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5617 AudioSystem::OutputDescriptor desc; 5618 void *param2 = NULL; 5619 5620 switch (event) { 5621 case AudioSystem::INPUT_OPENED: 5622 case AudioSystem::INPUT_CONFIG_CHANGED: 5623 desc.channels = mChannelMask; 5624 desc.samplingRate = mSampleRate; 5625 desc.format = mFormat; 5626 desc.frameCount = mFrameCount; 5627 desc.latency = 0; 5628 param2 = &desc; 5629 break; 5630 5631 case AudioSystem::INPUT_CLOSED: 5632 default: 5633 break; 5634 } 5635 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5636} 5637 5638void AudioFlinger::RecordThread::readInputParameters() 5639{ 5640 delete mRsmpInBuffer; 5641 // mRsmpInBuffer is always assigned a new[] below 5642 delete mRsmpOutBuffer; 5643 mRsmpOutBuffer = NULL; 5644 delete mResampler; 5645 mResampler = NULL; 5646 5647 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5648 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5649 mChannelCount = (uint16_t)popcount(mChannelMask); 5650 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5651 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5652 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5653 mFrameCount = mInputBytes / mFrameSize; 5654 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5655 5656 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5657 { 5658 int channelCount; 5659 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5660 // stereo to mono post process as the resampler always outputs stereo. 5661 if (mChannelCount == 1 && mReqChannelCount == 2) { 5662 channelCount = 1; 5663 } else { 5664 channelCount = 2; 5665 } 5666 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5667 mResampler->setSampleRate(mSampleRate); 5668 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5669 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5670 5671 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5672 if (mChannelCount == 1 && mReqChannelCount == 1) { 5673 mFrameCount >>= 1; 5674 } 5675 5676 } 5677 mRsmpInIndex = mFrameCount; 5678} 5679 5680unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5681{ 5682 Mutex::Autolock _l(mLock); 5683 if (initCheck() != NO_ERROR) { 5684 return 0; 5685 } 5686 5687 return mInput->stream->get_input_frames_lost(mInput->stream); 5688} 5689 5690uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5691{ 5692 Mutex::Autolock _l(mLock); 5693 uint32_t result = 0; 5694 if (getEffectChain_l(sessionId) != 0) { 5695 result = EFFECT_SESSION; 5696 } 5697 5698 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5699 result |= TRACK_SESSION; 5700 } 5701 5702 return result; 5703} 5704 5705AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5706{ 5707 Mutex::Autolock _l(mLock); 5708 return mTrack; 5709} 5710 5711AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5712{ 5713 Mutex::Autolock _l(mLock); 5714 return mInput; 5715} 5716 5717AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5718{ 5719 Mutex::Autolock _l(mLock); 5720 AudioStreamIn *input = mInput; 5721 mInput = NULL; 5722 return input; 5723} 5724 5725// this method must always be called either with ThreadBase mLock held or inside the thread loop 5726audio_stream_t* AudioFlinger::RecordThread::stream() const 5727{ 5728 if (mInput == NULL) { 5729 return NULL; 5730 } 5731 return &mInput->stream->common; 5732} 5733 5734 5735// ---------------------------------------------------------------------------- 5736 5737audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 5738{ 5739 if (!settingsAllowed()) { 5740 return 0; 5741 } 5742 Mutex::Autolock _l(mLock); 5743 return loadHwModule_l(name); 5744} 5745 5746// loadHwModule_l() must be called with AudioFlinger::mLock held 5747audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 5748{ 5749 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5750 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 5751 ALOGW("loadHwModule() module %s already loaded", name); 5752 return mAudioHwDevs.keyAt(i); 5753 } 5754 } 5755 5756 audio_hw_device_t *dev; 5757 5758 int rc = load_audio_interface(name, &dev); 5759 if (rc) { 5760 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 5761 return 0; 5762 } 5763 5764 mHardwareStatus = AUDIO_HW_INIT; 5765 rc = dev->init_check(dev); 5766 mHardwareStatus = AUDIO_HW_IDLE; 5767 if (rc) { 5768 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 5769 return 0; 5770 } 5771 5772 if ((mMasterVolumeSupportLvl != MVS_NONE) && 5773 (NULL != dev->set_master_volume)) { 5774 AutoMutex lock(mHardwareLock); 5775 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5776 dev->set_master_volume(dev, mMasterVolume); 5777 mHardwareStatus = AUDIO_HW_IDLE; 5778 } 5779 5780 audio_module_handle_t handle = nextUniqueId(); 5781 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 5782 5783 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 5784 name, dev->common.module->name, dev->common.module->id, handle); 5785 5786 return handle; 5787 5788} 5789 5790audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 5791 audio_devices_t *pDevices, 5792 uint32_t *pSamplingRate, 5793 audio_format_t *pFormat, 5794 audio_channel_mask_t *pChannelMask, 5795 uint32_t *pLatencyMs, 5796 audio_output_flags_t flags) 5797{ 5798 status_t status; 5799 PlaybackThread *thread = NULL; 5800 struct audio_config config = { 5801 sample_rate: pSamplingRate ? *pSamplingRate : 0, 5802 channel_mask: pChannelMask ? *pChannelMask : 0, 5803 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 5804 }; 5805 audio_stream_out_t *outStream = NULL; 5806 audio_hw_device_t *outHwDev; 5807 5808 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5809 module, 5810 (pDevices != NULL) ? (int)*pDevices : 0, 5811 config.sample_rate, 5812 config.format, 5813 config.channel_mask, 5814 flags); 5815 5816 if (pDevices == NULL || *pDevices == 0) { 5817 return 0; 5818 } 5819 5820 Mutex::Autolock _l(mLock); 5821 5822 outHwDev = findSuitableHwDev_l(module, *pDevices); 5823 if (outHwDev == NULL) 5824 return 0; 5825 5826 audio_io_handle_t id = nextUniqueId(); 5827 5828 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5829 5830 status = outHwDev->open_output_stream(outHwDev, 5831 id, 5832 *pDevices, 5833 (audio_output_flags_t)flags, 5834 &config, 5835 &outStream); 5836 5837 mHardwareStatus = AUDIO_HW_IDLE; 5838 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5839 outStream, 5840 config.sample_rate, 5841 config.format, 5842 config.channel_mask, 5843 status); 5844 5845 if (status == NO_ERROR && outStream != NULL) { 5846 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5847 5848 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 5849 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 5850 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 5851 thread = new DirectOutputThread(this, output, id, *pDevices); 5852 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5853 } else { 5854 thread = new MixerThread(this, output, id, *pDevices); 5855 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5856 } 5857 mPlaybackThreads.add(id, thread); 5858 5859 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 5860 if (pFormat != NULL) *pFormat = config.format; 5861 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 5862 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5863 5864 // notify client processes of the new output creation 5865 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5866 5867 // the first primary output opened designates the primary hw device 5868 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 5869 ALOGI("Using module %d has the primary audio interface", module); 5870 mPrimaryHardwareDev = outHwDev; 5871 5872 AutoMutex lock(mHardwareLock); 5873 mHardwareStatus = AUDIO_HW_SET_MODE; 5874 outHwDev->set_mode(outHwDev, mMode); 5875 5876 // Determine the level of master volume support the primary audio HAL has, 5877 // and set the initial master volume at the same time. 5878 float initialVolume = 1.0; 5879 mMasterVolumeSupportLvl = MVS_NONE; 5880 5881 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 5882 if ((NULL != outHwDev->get_master_volume) && 5883 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 5884 mMasterVolumeSupportLvl = MVS_FULL; 5885 } else { 5886 mMasterVolumeSupportLvl = MVS_SETONLY; 5887 initialVolume = 1.0; 5888 } 5889 5890 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5891 if ((NULL == outHwDev->set_master_volume) || 5892 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 5893 mMasterVolumeSupportLvl = MVS_NONE; 5894 } 5895 // now that we have a primary device, initialize master volume on other devices 5896 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5897 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 5898 5899 if ((dev != mPrimaryHardwareDev) && 5900 (NULL != dev->set_master_volume)) { 5901 dev->set_master_volume(dev, initialVolume); 5902 } 5903 } 5904 mHardwareStatus = AUDIO_HW_IDLE; 5905 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 5906 ? initialVolume 5907 : 1.0; 5908 mMasterVolume = initialVolume; 5909 } 5910 return id; 5911 } 5912 5913 return 0; 5914} 5915 5916audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5917 audio_io_handle_t output2) 5918{ 5919 Mutex::Autolock _l(mLock); 5920 MixerThread *thread1 = checkMixerThread_l(output1); 5921 MixerThread *thread2 = checkMixerThread_l(output2); 5922 5923 if (thread1 == NULL || thread2 == NULL) { 5924 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5925 return 0; 5926 } 5927 5928 audio_io_handle_t id = nextUniqueId(); 5929 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5930 thread->addOutputTrack(thread2); 5931 mPlaybackThreads.add(id, thread); 5932 // notify client processes of the new output creation 5933 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5934 return id; 5935} 5936 5937status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5938{ 5939 // keep strong reference on the playback thread so that 5940 // it is not destroyed while exit() is executed 5941 sp<PlaybackThread> thread; 5942 { 5943 Mutex::Autolock _l(mLock); 5944 thread = checkPlaybackThread_l(output); 5945 if (thread == NULL) { 5946 return BAD_VALUE; 5947 } 5948 5949 ALOGV("closeOutput() %d", output); 5950 5951 if (thread->type() == ThreadBase::MIXER) { 5952 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5953 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5954 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5955 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5956 } 5957 } 5958 } 5959 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5960 mPlaybackThreads.removeItem(output); 5961 } 5962 thread->exit(); 5963 // The thread entity (active unit of execution) is no longer running here, 5964 // but the ThreadBase container still exists. 5965 5966 if (thread->type() != ThreadBase::DUPLICATING) { 5967 AudioStreamOut *out = thread->clearOutput(); 5968 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5969 // from now on thread->mOutput is NULL 5970 out->hwDev->close_output_stream(out->hwDev, out->stream); 5971 delete out; 5972 } 5973 return NO_ERROR; 5974} 5975 5976status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5977{ 5978 Mutex::Autolock _l(mLock); 5979 PlaybackThread *thread = checkPlaybackThread_l(output); 5980 5981 if (thread == NULL) { 5982 return BAD_VALUE; 5983 } 5984 5985 ALOGV("suspendOutput() %d", output); 5986 thread->suspend(); 5987 5988 return NO_ERROR; 5989} 5990 5991status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5992{ 5993 Mutex::Autolock _l(mLock); 5994 PlaybackThread *thread = checkPlaybackThread_l(output); 5995 5996 if (thread == NULL) { 5997 return BAD_VALUE; 5998 } 5999 6000 ALOGV("restoreOutput() %d", output); 6001 6002 thread->restore(); 6003 6004 return NO_ERROR; 6005} 6006 6007audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6008 audio_devices_t *pDevices, 6009 uint32_t *pSamplingRate, 6010 audio_format_t *pFormat, 6011 uint32_t *pChannelMask) 6012{ 6013 status_t status; 6014 RecordThread *thread = NULL; 6015 struct audio_config config = { 6016 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6017 channel_mask: pChannelMask ? *pChannelMask : 0, 6018 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6019 }; 6020 uint32_t reqSamplingRate = config.sample_rate; 6021 audio_format_t reqFormat = config.format; 6022 audio_channel_mask_t reqChannels = config.channel_mask; 6023 audio_stream_in_t *inStream = NULL; 6024 audio_hw_device_t *inHwDev; 6025 6026 if (pDevices == NULL || *pDevices == 0) { 6027 return 0; 6028 } 6029 6030 Mutex::Autolock _l(mLock); 6031 6032 inHwDev = findSuitableHwDev_l(module, *pDevices); 6033 if (inHwDev == NULL) 6034 return 0; 6035 6036 audio_io_handle_t id = nextUniqueId(); 6037 6038 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6039 &inStream); 6040 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6041 inStream, 6042 config.sample_rate, 6043 config.format, 6044 config.channel_mask, 6045 status); 6046 6047 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6048 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6049 // or stereo to mono conversions on 16 bit PCM inputs. 6050 if (status == BAD_VALUE && 6051 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6052 (config.sample_rate <= 2 * reqSamplingRate) && 6053 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6054 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6055 inStream = NULL; 6056 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6057 } 6058 6059 if (status == NO_ERROR && inStream != NULL) { 6060 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6061 6062 // Start record thread 6063 // RecorThread require both input and output device indication to forward to audio 6064 // pre processing modules 6065 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6066 thread = new RecordThread(this, 6067 input, 6068 reqSamplingRate, 6069 reqChannels, 6070 id, 6071 device); 6072 mRecordThreads.add(id, thread); 6073 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6074 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6075 if (pFormat != NULL) *pFormat = config.format; 6076 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6077 6078 input->stream->common.standby(&input->stream->common); 6079 6080 // notify client processes of the new input creation 6081 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6082 return id; 6083 } 6084 6085 return 0; 6086} 6087 6088status_t AudioFlinger::closeInput(audio_io_handle_t input) 6089{ 6090 // keep strong reference on the record thread so that 6091 // it is not destroyed while exit() is executed 6092 sp<RecordThread> thread; 6093 { 6094 Mutex::Autolock _l(mLock); 6095 thread = checkRecordThread_l(input); 6096 if (thread == NULL) { 6097 return BAD_VALUE; 6098 } 6099 6100 ALOGV("closeInput() %d", input); 6101 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6102 mRecordThreads.removeItem(input); 6103 } 6104 thread->exit(); 6105 // The thread entity (active unit of execution) is no longer running here, 6106 // but the ThreadBase container still exists. 6107 6108 AudioStreamIn *in = thread->clearInput(); 6109 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6110 // from now on thread->mInput is NULL 6111 in->hwDev->close_input_stream(in->hwDev, in->stream); 6112 delete in; 6113 6114 return NO_ERROR; 6115} 6116 6117status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6118{ 6119 Mutex::Autolock _l(mLock); 6120 MixerThread *dstThread = checkMixerThread_l(output); 6121 if (dstThread == NULL) { 6122 ALOGW("setStreamOutput() bad output id %d", output); 6123 return BAD_VALUE; 6124 } 6125 6126 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6127 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6128 6129 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6130 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6131 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6132 MixerThread *srcThread = (MixerThread *)thread; 6133 srcThread->invalidateTracks(stream); 6134 } 6135 } 6136 6137 return NO_ERROR; 6138} 6139 6140 6141int AudioFlinger::newAudioSessionId() 6142{ 6143 return nextUniqueId(); 6144} 6145 6146void AudioFlinger::acquireAudioSessionId(int audioSession) 6147{ 6148 Mutex::Autolock _l(mLock); 6149 pid_t caller = IPCThreadState::self()->getCallingPid(); 6150 ALOGV("acquiring %d from %d", audioSession, caller); 6151 size_t num = mAudioSessionRefs.size(); 6152 for (size_t i = 0; i< num; i++) { 6153 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6154 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6155 ref->mCnt++; 6156 ALOGV(" incremented refcount to %d", ref->mCnt); 6157 return; 6158 } 6159 } 6160 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6161 ALOGV(" added new entry for %d", audioSession); 6162} 6163 6164void AudioFlinger::releaseAudioSessionId(int audioSession) 6165{ 6166 Mutex::Autolock _l(mLock); 6167 pid_t caller = IPCThreadState::self()->getCallingPid(); 6168 ALOGV("releasing %d from %d", audioSession, caller); 6169 size_t num = mAudioSessionRefs.size(); 6170 for (size_t i = 0; i< num; i++) { 6171 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6172 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6173 ref->mCnt--; 6174 ALOGV(" decremented refcount to %d", ref->mCnt); 6175 if (ref->mCnt == 0) { 6176 mAudioSessionRefs.removeAt(i); 6177 delete ref; 6178 purgeStaleEffects_l(); 6179 } 6180 return; 6181 } 6182 } 6183 ALOGW("session id %d not found for pid %d", audioSession, caller); 6184} 6185 6186void AudioFlinger::purgeStaleEffects_l() { 6187 6188 ALOGV("purging stale effects"); 6189 6190 Vector< sp<EffectChain> > chains; 6191 6192 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6193 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6194 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6195 sp<EffectChain> ec = t->mEffectChains[j]; 6196 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6197 chains.push(ec); 6198 } 6199 } 6200 } 6201 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6202 sp<RecordThread> t = mRecordThreads.valueAt(i); 6203 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6204 sp<EffectChain> ec = t->mEffectChains[j]; 6205 chains.push(ec); 6206 } 6207 } 6208 6209 for (size_t i = 0; i < chains.size(); i++) { 6210 sp<EffectChain> ec = chains[i]; 6211 int sessionid = ec->sessionId(); 6212 sp<ThreadBase> t = ec->mThread.promote(); 6213 if (t == 0) { 6214 continue; 6215 } 6216 size_t numsessionrefs = mAudioSessionRefs.size(); 6217 bool found = false; 6218 for (size_t k = 0; k < numsessionrefs; k++) { 6219 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6220 if (ref->mSessionid == sessionid) { 6221 ALOGV(" session %d still exists for %d with %d refs", 6222 sessionid, ref->mPid, ref->mCnt); 6223 found = true; 6224 break; 6225 } 6226 } 6227 if (!found) { 6228 // remove all effects from the chain 6229 while (ec->mEffects.size()) { 6230 sp<EffectModule> effect = ec->mEffects[0]; 6231 effect->unPin(); 6232 Mutex::Autolock _l (t->mLock); 6233 t->removeEffect_l(effect); 6234 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6235 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6236 if (handle != 0) { 6237 handle->mEffect.clear(); 6238 if (handle->mHasControl && handle->mEnabled) { 6239 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6240 } 6241 } 6242 } 6243 AudioSystem::unregisterEffect(effect->id()); 6244 } 6245 } 6246 } 6247 return; 6248} 6249 6250// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6251AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6252{ 6253 return mPlaybackThreads.valueFor(output).get(); 6254} 6255 6256// checkMixerThread_l() must be called with AudioFlinger::mLock held 6257AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6258{ 6259 PlaybackThread *thread = checkPlaybackThread_l(output); 6260 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6261} 6262 6263// checkRecordThread_l() must be called with AudioFlinger::mLock held 6264AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6265{ 6266 return mRecordThreads.valueFor(input).get(); 6267} 6268 6269uint32_t AudioFlinger::nextUniqueId() 6270{ 6271 return android_atomic_inc(&mNextUniqueId); 6272} 6273 6274AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6275{ 6276 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6277 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6278 AudioStreamOut *output = thread->getOutput(); 6279 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6280 return thread; 6281 } 6282 } 6283 return NULL; 6284} 6285 6286uint32_t AudioFlinger::primaryOutputDevice_l() const 6287{ 6288 PlaybackThread *thread = primaryPlaybackThread_l(); 6289 6290 if (thread == NULL) { 6291 return 0; 6292 } 6293 6294 return thread->device(); 6295} 6296 6297sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6298 int triggerSession, 6299 int listenerSession, 6300 sync_event_callback_t callBack, 6301 void *cookie) 6302{ 6303 Mutex::Autolock _l(mLock); 6304 6305 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6306 status_t playStatus = NAME_NOT_FOUND; 6307 status_t recStatus = NAME_NOT_FOUND; 6308 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6309 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6310 if (playStatus == NO_ERROR) { 6311 return event; 6312 } 6313 } 6314 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6315 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6316 if (recStatus == NO_ERROR) { 6317 return event; 6318 } 6319 } 6320 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6321 mPendingSyncEvents.add(event); 6322 } else { 6323 ALOGV("createSyncEvent() invalid event %d", event->type()); 6324 event.clear(); 6325 } 6326 return event; 6327} 6328 6329// ---------------------------------------------------------------------------- 6330// Effect management 6331// ---------------------------------------------------------------------------- 6332 6333 6334status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6335{ 6336 Mutex::Autolock _l(mLock); 6337 return EffectQueryNumberEffects(numEffects); 6338} 6339 6340status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6341{ 6342 Mutex::Autolock _l(mLock); 6343 return EffectQueryEffect(index, descriptor); 6344} 6345 6346status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6347 effect_descriptor_t *descriptor) const 6348{ 6349 Mutex::Autolock _l(mLock); 6350 return EffectGetDescriptor(pUuid, descriptor); 6351} 6352 6353 6354sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6355 effect_descriptor_t *pDesc, 6356 const sp<IEffectClient>& effectClient, 6357 int32_t priority, 6358 audio_io_handle_t io, 6359 int sessionId, 6360 status_t *status, 6361 int *id, 6362 int *enabled) 6363{ 6364 status_t lStatus = NO_ERROR; 6365 sp<EffectHandle> handle; 6366 effect_descriptor_t desc; 6367 6368 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6369 pid, effectClient.get(), priority, sessionId, io); 6370 6371 if (pDesc == NULL) { 6372 lStatus = BAD_VALUE; 6373 goto Exit; 6374 } 6375 6376 // check audio settings permission for global effects 6377 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6378 lStatus = PERMISSION_DENIED; 6379 goto Exit; 6380 } 6381 6382 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6383 // that can only be created by audio policy manager (running in same process) 6384 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6385 lStatus = PERMISSION_DENIED; 6386 goto Exit; 6387 } 6388 6389 if (io == 0) { 6390 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6391 // output must be specified by AudioPolicyManager when using session 6392 // AUDIO_SESSION_OUTPUT_STAGE 6393 lStatus = BAD_VALUE; 6394 goto Exit; 6395 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6396 // if the output returned by getOutputForEffect() is removed before we lock the 6397 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6398 // and we will exit safely 6399 io = AudioSystem::getOutputForEffect(&desc); 6400 } 6401 } 6402 6403 { 6404 Mutex::Autolock _l(mLock); 6405 6406 6407 if (!EffectIsNullUuid(&pDesc->uuid)) { 6408 // if uuid is specified, request effect descriptor 6409 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6410 if (lStatus < 0) { 6411 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6412 goto Exit; 6413 } 6414 } else { 6415 // if uuid is not specified, look for an available implementation 6416 // of the required type in effect factory 6417 if (EffectIsNullUuid(&pDesc->type)) { 6418 ALOGW("createEffect() no effect type"); 6419 lStatus = BAD_VALUE; 6420 goto Exit; 6421 } 6422 uint32_t numEffects = 0; 6423 effect_descriptor_t d; 6424 d.flags = 0; // prevent compiler warning 6425 bool found = false; 6426 6427 lStatus = EffectQueryNumberEffects(&numEffects); 6428 if (lStatus < 0) { 6429 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6430 goto Exit; 6431 } 6432 for (uint32_t i = 0; i < numEffects; i++) { 6433 lStatus = EffectQueryEffect(i, &desc); 6434 if (lStatus < 0) { 6435 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6436 continue; 6437 } 6438 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6439 // If matching type found save effect descriptor. If the session is 6440 // 0 and the effect is not auxiliary, continue enumeration in case 6441 // an auxiliary version of this effect type is available 6442 found = true; 6443 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6444 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6445 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6446 break; 6447 } 6448 } 6449 } 6450 if (!found) { 6451 lStatus = BAD_VALUE; 6452 ALOGW("createEffect() effect not found"); 6453 goto Exit; 6454 } 6455 // For same effect type, chose auxiliary version over insert version if 6456 // connect to output mix (Compliance to OpenSL ES) 6457 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6458 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6459 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6460 } 6461 } 6462 6463 // Do not allow auxiliary effects on a session different from 0 (output mix) 6464 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6465 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6466 lStatus = INVALID_OPERATION; 6467 goto Exit; 6468 } 6469 6470 // check recording permission for visualizer 6471 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6472 !recordingAllowed()) { 6473 lStatus = PERMISSION_DENIED; 6474 goto Exit; 6475 } 6476 6477 // return effect descriptor 6478 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6479 6480 // If output is not specified try to find a matching audio session ID in one of the 6481 // output threads. 6482 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6483 // because of code checking output when entering the function. 6484 // Note: io is never 0 when creating an effect on an input 6485 if (io == 0) { 6486 // look for the thread where the specified audio session is present 6487 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6488 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6489 io = mPlaybackThreads.keyAt(i); 6490 break; 6491 } 6492 } 6493 if (io == 0) { 6494 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6495 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6496 io = mRecordThreads.keyAt(i); 6497 break; 6498 } 6499 } 6500 } 6501 // If no output thread contains the requested session ID, default to 6502 // first output. The effect chain will be moved to the correct output 6503 // thread when a track with the same session ID is created 6504 if (io == 0 && mPlaybackThreads.size()) { 6505 io = mPlaybackThreads.keyAt(0); 6506 } 6507 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6508 } 6509 ThreadBase *thread = checkRecordThread_l(io); 6510 if (thread == NULL) { 6511 thread = checkPlaybackThread_l(io); 6512 if (thread == NULL) { 6513 ALOGE("createEffect() unknown output thread"); 6514 lStatus = BAD_VALUE; 6515 goto Exit; 6516 } 6517 } 6518 6519 sp<Client> client = registerPid_l(pid); 6520 6521 // create effect on selected output thread 6522 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6523 &desc, enabled, &lStatus); 6524 if (handle != 0 && id != NULL) { 6525 *id = handle->id(); 6526 } 6527 } 6528 6529Exit: 6530 if (status != NULL) { 6531 *status = lStatus; 6532 } 6533 return handle; 6534} 6535 6536status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6537 audio_io_handle_t dstOutput) 6538{ 6539 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6540 sessionId, srcOutput, dstOutput); 6541 Mutex::Autolock _l(mLock); 6542 if (srcOutput == dstOutput) { 6543 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6544 return NO_ERROR; 6545 } 6546 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6547 if (srcThread == NULL) { 6548 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6549 return BAD_VALUE; 6550 } 6551 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6552 if (dstThread == NULL) { 6553 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6554 return BAD_VALUE; 6555 } 6556 6557 Mutex::Autolock _dl(dstThread->mLock); 6558 Mutex::Autolock _sl(srcThread->mLock); 6559 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6560 6561 return NO_ERROR; 6562} 6563 6564// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6565status_t AudioFlinger::moveEffectChain_l(int sessionId, 6566 AudioFlinger::PlaybackThread *srcThread, 6567 AudioFlinger::PlaybackThread *dstThread, 6568 bool reRegister) 6569{ 6570 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6571 sessionId, srcThread, dstThread); 6572 6573 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6574 if (chain == 0) { 6575 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6576 sessionId, srcThread); 6577 return INVALID_OPERATION; 6578 } 6579 6580 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6581 // so that a new chain is created with correct parameters when first effect is added. This is 6582 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6583 // removed. 6584 srcThread->removeEffectChain_l(chain); 6585 6586 // transfer all effects one by one so that new effect chain is created on new thread with 6587 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6588 audio_io_handle_t dstOutput = dstThread->id(); 6589 sp<EffectChain> dstChain; 6590 uint32_t strategy = 0; // prevent compiler warning 6591 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6592 while (effect != 0) { 6593 srcThread->removeEffect_l(effect); 6594 dstThread->addEffect_l(effect); 6595 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6596 if (effect->state() == EffectModule::ACTIVE || 6597 effect->state() == EffectModule::STOPPING) { 6598 effect->start(); 6599 } 6600 // if the move request is not received from audio policy manager, the effect must be 6601 // re-registered with the new strategy and output 6602 if (dstChain == 0) { 6603 dstChain = effect->chain().promote(); 6604 if (dstChain == 0) { 6605 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6606 srcThread->addEffect_l(effect); 6607 return NO_INIT; 6608 } 6609 strategy = dstChain->strategy(); 6610 } 6611 if (reRegister) { 6612 AudioSystem::unregisterEffect(effect->id()); 6613 AudioSystem::registerEffect(&effect->desc(), 6614 dstOutput, 6615 strategy, 6616 sessionId, 6617 effect->id()); 6618 } 6619 effect = chain->getEffectFromId_l(0); 6620 } 6621 6622 return NO_ERROR; 6623} 6624 6625 6626// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6627sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6628 const sp<AudioFlinger::Client>& client, 6629 const sp<IEffectClient>& effectClient, 6630 int32_t priority, 6631 int sessionId, 6632 effect_descriptor_t *desc, 6633 int *enabled, 6634 status_t *status 6635 ) 6636{ 6637 sp<EffectModule> effect; 6638 sp<EffectHandle> handle; 6639 status_t lStatus; 6640 sp<EffectChain> chain; 6641 bool chainCreated = false; 6642 bool effectCreated = false; 6643 bool effectRegistered = false; 6644 6645 lStatus = initCheck(); 6646 if (lStatus != NO_ERROR) { 6647 ALOGW("createEffect_l() Audio driver not initialized."); 6648 goto Exit; 6649 } 6650 6651 // Do not allow effects with session ID 0 on direct output or duplicating threads 6652 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6653 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6654 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6655 desc->name, sessionId); 6656 lStatus = BAD_VALUE; 6657 goto Exit; 6658 } 6659 // Only Pre processor effects are allowed on input threads and only on input threads 6660 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6661 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6662 desc->name, desc->flags, mType); 6663 lStatus = BAD_VALUE; 6664 goto Exit; 6665 } 6666 6667 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6668 6669 { // scope for mLock 6670 Mutex::Autolock _l(mLock); 6671 6672 // check for existing effect chain with the requested audio session 6673 chain = getEffectChain_l(sessionId); 6674 if (chain == 0) { 6675 // create a new chain for this session 6676 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6677 chain = new EffectChain(this, sessionId); 6678 addEffectChain_l(chain); 6679 chain->setStrategy(getStrategyForSession_l(sessionId)); 6680 chainCreated = true; 6681 } else { 6682 effect = chain->getEffectFromDesc_l(desc); 6683 } 6684 6685 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6686 6687 if (effect == 0) { 6688 int id = mAudioFlinger->nextUniqueId(); 6689 // Check CPU and memory usage 6690 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6691 if (lStatus != NO_ERROR) { 6692 goto Exit; 6693 } 6694 effectRegistered = true; 6695 // create a new effect module if none present in the chain 6696 effect = new EffectModule(this, chain, desc, id, sessionId); 6697 lStatus = effect->status(); 6698 if (lStatus != NO_ERROR) { 6699 goto Exit; 6700 } 6701 lStatus = chain->addEffect_l(effect); 6702 if (lStatus != NO_ERROR) { 6703 goto Exit; 6704 } 6705 effectCreated = true; 6706 6707 effect->setDevice(mDevice); 6708 effect->setMode(mAudioFlinger->getMode()); 6709 } 6710 // create effect handle and connect it to effect module 6711 handle = new EffectHandle(effect, client, effectClient, priority); 6712 lStatus = effect->addHandle(handle); 6713 if (enabled != NULL) { 6714 *enabled = (int)effect->isEnabled(); 6715 } 6716 } 6717 6718Exit: 6719 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6720 Mutex::Autolock _l(mLock); 6721 if (effectCreated) { 6722 chain->removeEffect_l(effect); 6723 } 6724 if (effectRegistered) { 6725 AudioSystem::unregisterEffect(effect->id()); 6726 } 6727 if (chainCreated) { 6728 removeEffectChain_l(chain); 6729 } 6730 handle.clear(); 6731 } 6732 6733 if (status != NULL) { 6734 *status = lStatus; 6735 } 6736 return handle; 6737} 6738 6739sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6740{ 6741 sp<EffectChain> chain = getEffectChain_l(sessionId); 6742 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6743} 6744 6745// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6746// PlaybackThread::mLock held 6747status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6748{ 6749 // check for existing effect chain with the requested audio session 6750 int sessionId = effect->sessionId(); 6751 sp<EffectChain> chain = getEffectChain_l(sessionId); 6752 bool chainCreated = false; 6753 6754 if (chain == 0) { 6755 // create a new chain for this session 6756 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6757 chain = new EffectChain(this, sessionId); 6758 addEffectChain_l(chain); 6759 chain->setStrategy(getStrategyForSession_l(sessionId)); 6760 chainCreated = true; 6761 } 6762 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6763 6764 if (chain->getEffectFromId_l(effect->id()) != 0) { 6765 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6766 this, effect->desc().name, chain.get()); 6767 return BAD_VALUE; 6768 } 6769 6770 status_t status = chain->addEffect_l(effect); 6771 if (status != NO_ERROR) { 6772 if (chainCreated) { 6773 removeEffectChain_l(chain); 6774 } 6775 return status; 6776 } 6777 6778 effect->setDevice(mDevice); 6779 effect->setMode(mAudioFlinger->getMode()); 6780 return NO_ERROR; 6781} 6782 6783void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6784 6785 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6786 effect_descriptor_t desc = effect->desc(); 6787 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6788 detachAuxEffect_l(effect->id()); 6789 } 6790 6791 sp<EffectChain> chain = effect->chain().promote(); 6792 if (chain != 0) { 6793 // remove effect chain if removing last effect 6794 if (chain->removeEffect_l(effect) == 0) { 6795 removeEffectChain_l(chain); 6796 } 6797 } else { 6798 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6799 } 6800} 6801 6802void AudioFlinger::ThreadBase::lockEffectChains_l( 6803 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6804{ 6805 effectChains = mEffectChains; 6806 for (size_t i = 0; i < mEffectChains.size(); i++) { 6807 mEffectChains[i]->lock(); 6808 } 6809} 6810 6811void AudioFlinger::ThreadBase::unlockEffectChains( 6812 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6813{ 6814 for (size_t i = 0; i < effectChains.size(); i++) { 6815 effectChains[i]->unlock(); 6816 } 6817} 6818 6819sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6820{ 6821 Mutex::Autolock _l(mLock); 6822 return getEffectChain_l(sessionId); 6823} 6824 6825sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6826{ 6827 size_t size = mEffectChains.size(); 6828 for (size_t i = 0; i < size; i++) { 6829 if (mEffectChains[i]->sessionId() == sessionId) { 6830 return mEffectChains[i]; 6831 } 6832 } 6833 return 0; 6834} 6835 6836void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6837{ 6838 Mutex::Autolock _l(mLock); 6839 size_t size = mEffectChains.size(); 6840 for (size_t i = 0; i < size; i++) { 6841 mEffectChains[i]->setMode_l(mode); 6842 } 6843} 6844 6845void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6846 const wp<EffectHandle>& handle, 6847 bool unpinIfLast) { 6848 6849 Mutex::Autolock _l(mLock); 6850 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6851 // delete the effect module if removing last handle on it 6852 if (effect->removeHandle(handle) == 0) { 6853 if (!effect->isPinned() || unpinIfLast) { 6854 removeEffect_l(effect); 6855 AudioSystem::unregisterEffect(effect->id()); 6856 } 6857 } 6858} 6859 6860status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6861{ 6862 int session = chain->sessionId(); 6863 int16_t *buffer = mMixBuffer; 6864 bool ownsBuffer = false; 6865 6866 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6867 if (session > 0) { 6868 // Only one effect chain can be present in direct output thread and it uses 6869 // the mix buffer as input 6870 if (mType != DIRECT) { 6871 size_t numSamples = mFrameCount * mChannelCount; 6872 buffer = new int16_t[numSamples]; 6873 memset(buffer, 0, numSamples * sizeof(int16_t)); 6874 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6875 ownsBuffer = true; 6876 } 6877 6878 // Attach all tracks with same session ID to this chain. 6879 for (size_t i = 0; i < mTracks.size(); ++i) { 6880 sp<Track> track = mTracks[i]; 6881 if (session == track->sessionId()) { 6882 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6883 track->setMainBuffer(buffer); 6884 chain->incTrackCnt(); 6885 } 6886 } 6887 6888 // indicate all active tracks in the chain 6889 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6890 sp<Track> track = mActiveTracks[i].promote(); 6891 if (track == 0) continue; 6892 if (session == track->sessionId()) { 6893 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6894 chain->incActiveTrackCnt(); 6895 } 6896 } 6897 } 6898 6899 chain->setInBuffer(buffer, ownsBuffer); 6900 chain->setOutBuffer(mMixBuffer); 6901 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6902 // chains list in order to be processed last as it contains output stage effects 6903 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6904 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6905 // after track specific effects and before output stage 6906 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6907 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6908 // Effect chain for other sessions are inserted at beginning of effect 6909 // chains list to be processed before output mix effects. Relative order between other 6910 // sessions is not important 6911 size_t size = mEffectChains.size(); 6912 size_t i = 0; 6913 for (i = 0; i < size; i++) { 6914 if (mEffectChains[i]->sessionId() < session) break; 6915 } 6916 mEffectChains.insertAt(chain, i); 6917 checkSuspendOnAddEffectChain_l(chain); 6918 6919 return NO_ERROR; 6920} 6921 6922size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6923{ 6924 int session = chain->sessionId(); 6925 6926 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6927 6928 for (size_t i = 0; i < mEffectChains.size(); i++) { 6929 if (chain == mEffectChains[i]) { 6930 mEffectChains.removeAt(i); 6931 // detach all active tracks from the chain 6932 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6933 sp<Track> track = mActiveTracks[i].promote(); 6934 if (track == 0) continue; 6935 if (session == track->sessionId()) { 6936 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6937 chain.get(), session); 6938 chain->decActiveTrackCnt(); 6939 } 6940 } 6941 6942 // detach all tracks with same session ID from this chain 6943 for (size_t i = 0; i < mTracks.size(); ++i) { 6944 sp<Track> track = mTracks[i]; 6945 if (session == track->sessionId()) { 6946 track->setMainBuffer(mMixBuffer); 6947 chain->decTrackCnt(); 6948 } 6949 } 6950 break; 6951 } 6952 } 6953 return mEffectChains.size(); 6954} 6955 6956status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6957 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6958{ 6959 Mutex::Autolock _l(mLock); 6960 return attachAuxEffect_l(track, EffectId); 6961} 6962 6963status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6964 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6965{ 6966 status_t status = NO_ERROR; 6967 6968 if (EffectId == 0) { 6969 track->setAuxBuffer(0, NULL); 6970 } else { 6971 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6972 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6973 if (effect != 0) { 6974 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6975 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6976 } else { 6977 status = INVALID_OPERATION; 6978 } 6979 } else { 6980 status = BAD_VALUE; 6981 } 6982 } 6983 return status; 6984} 6985 6986void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6987{ 6988 for (size_t i = 0; i < mTracks.size(); ++i) { 6989 sp<Track> track = mTracks[i]; 6990 if (track->auxEffectId() == effectId) { 6991 attachAuxEffect_l(track, 0); 6992 } 6993 } 6994} 6995 6996status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6997{ 6998 // only one chain per input thread 6999 if (mEffectChains.size() != 0) { 7000 return INVALID_OPERATION; 7001 } 7002 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7003 7004 chain->setInBuffer(NULL); 7005 chain->setOutBuffer(NULL); 7006 7007 checkSuspendOnAddEffectChain_l(chain); 7008 7009 mEffectChains.add(chain); 7010 7011 return NO_ERROR; 7012} 7013 7014size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7015{ 7016 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7017 ALOGW_IF(mEffectChains.size() != 1, 7018 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7019 chain.get(), mEffectChains.size(), this); 7020 if (mEffectChains.size() == 1) { 7021 mEffectChains.removeAt(0); 7022 } 7023 return 0; 7024} 7025 7026// ---------------------------------------------------------------------------- 7027// EffectModule implementation 7028// ---------------------------------------------------------------------------- 7029 7030#undef LOG_TAG 7031#define LOG_TAG "AudioFlinger::EffectModule" 7032 7033AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7034 const wp<AudioFlinger::EffectChain>& chain, 7035 effect_descriptor_t *desc, 7036 int id, 7037 int sessionId) 7038 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7039 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7040{ 7041 ALOGV("Constructor %p", this); 7042 int lStatus; 7043 if (thread == NULL) { 7044 return; 7045 } 7046 7047 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7048 7049 // create effect engine from effect factory 7050 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7051 7052 if (mStatus != NO_ERROR) { 7053 return; 7054 } 7055 lStatus = init(); 7056 if (lStatus < 0) { 7057 mStatus = lStatus; 7058 goto Error; 7059 } 7060 7061 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7062 mPinned = true; 7063 } 7064 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7065 return; 7066Error: 7067 EffectRelease(mEffectInterface); 7068 mEffectInterface = NULL; 7069 ALOGV("Constructor Error %d", mStatus); 7070} 7071 7072AudioFlinger::EffectModule::~EffectModule() 7073{ 7074 ALOGV("Destructor %p", this); 7075 if (mEffectInterface != NULL) { 7076 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7077 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7078 sp<ThreadBase> thread = mThread.promote(); 7079 if (thread != 0) { 7080 audio_stream_t *stream = thread->stream(); 7081 if (stream != NULL) { 7082 stream->remove_audio_effect(stream, mEffectInterface); 7083 } 7084 } 7085 } 7086 // release effect engine 7087 EffectRelease(mEffectInterface); 7088 } 7089} 7090 7091status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7092{ 7093 status_t status; 7094 7095 Mutex::Autolock _l(mLock); 7096 int priority = handle->priority(); 7097 size_t size = mHandles.size(); 7098 sp<EffectHandle> h; 7099 size_t i; 7100 for (i = 0; i < size; i++) { 7101 h = mHandles[i].promote(); 7102 if (h == 0) continue; 7103 if (h->priority() <= priority) break; 7104 } 7105 // if inserted in first place, move effect control from previous owner to this handle 7106 if (i == 0) { 7107 bool enabled = false; 7108 if (h != 0) { 7109 enabled = h->enabled(); 7110 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7111 } 7112 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7113 status = NO_ERROR; 7114 } else { 7115 status = ALREADY_EXISTS; 7116 } 7117 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7118 mHandles.insertAt(handle, i); 7119 return status; 7120} 7121 7122size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7123{ 7124 Mutex::Autolock _l(mLock); 7125 size_t size = mHandles.size(); 7126 size_t i; 7127 for (i = 0; i < size; i++) { 7128 if (mHandles[i] == handle) break; 7129 } 7130 if (i == size) { 7131 return size; 7132 } 7133 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7134 7135 bool enabled = false; 7136 EffectHandle *hdl = handle.unsafe_get(); 7137 if (hdl != NULL) { 7138 ALOGV("removeHandle() unsafe_get OK"); 7139 enabled = hdl->enabled(); 7140 } 7141 mHandles.removeAt(i); 7142 size = mHandles.size(); 7143 // if removed from first place, move effect control from this handle to next in line 7144 if (i == 0 && size != 0) { 7145 sp<EffectHandle> h = mHandles[0].promote(); 7146 if (h != 0) { 7147 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7148 } 7149 } 7150 7151 // Prevent calls to process() and other functions on effect interface from now on. 7152 // The effect engine will be released by the destructor when the last strong reference on 7153 // this object is released which can happen after next process is called. 7154 if (size == 0 && !mPinned) { 7155 mState = DESTROYED; 7156 } 7157 7158 return size; 7159} 7160 7161sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7162{ 7163 Mutex::Autolock _l(mLock); 7164 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7165} 7166 7167void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7168{ 7169 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7170 // keep a strong reference on this EffectModule to avoid calling the 7171 // destructor before we exit 7172 sp<EffectModule> keep(this); 7173 { 7174 sp<ThreadBase> thread = mThread.promote(); 7175 if (thread != 0) { 7176 thread->disconnectEffect(keep, handle, unpinIfLast); 7177 } 7178 } 7179} 7180 7181void AudioFlinger::EffectModule::updateState() { 7182 Mutex::Autolock _l(mLock); 7183 7184 switch (mState) { 7185 case RESTART: 7186 reset_l(); 7187 // FALL THROUGH 7188 7189 case STARTING: 7190 // clear auxiliary effect input buffer for next accumulation 7191 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7192 memset(mConfig.inputCfg.buffer.raw, 7193 0, 7194 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7195 } 7196 start_l(); 7197 mState = ACTIVE; 7198 break; 7199 case STOPPING: 7200 stop_l(); 7201 mDisableWaitCnt = mMaxDisableWaitCnt; 7202 mState = STOPPED; 7203 break; 7204 case STOPPED: 7205 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7206 // turn off sequence. 7207 if (--mDisableWaitCnt == 0) { 7208 reset_l(); 7209 mState = IDLE; 7210 } 7211 break; 7212 default: //IDLE , ACTIVE, DESTROYED 7213 break; 7214 } 7215} 7216 7217void AudioFlinger::EffectModule::process() 7218{ 7219 Mutex::Autolock _l(mLock); 7220 7221 if (mState == DESTROYED || mEffectInterface == NULL || 7222 mConfig.inputCfg.buffer.raw == NULL || 7223 mConfig.outputCfg.buffer.raw == NULL) { 7224 return; 7225 } 7226 7227 if (isProcessEnabled()) { 7228 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7229 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7230 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7231 mConfig.inputCfg.buffer.s32, 7232 mConfig.inputCfg.buffer.frameCount/2); 7233 } 7234 7235 // do the actual processing in the effect engine 7236 int ret = (*mEffectInterface)->process(mEffectInterface, 7237 &mConfig.inputCfg.buffer, 7238 &mConfig.outputCfg.buffer); 7239 7240 // force transition to IDLE state when engine is ready 7241 if (mState == STOPPED && ret == -ENODATA) { 7242 mDisableWaitCnt = 1; 7243 } 7244 7245 // clear auxiliary effect input buffer for next accumulation 7246 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7247 memset(mConfig.inputCfg.buffer.raw, 0, 7248 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7249 } 7250 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7251 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7252 // If an insert effect is idle and input buffer is different from output buffer, 7253 // accumulate input onto output 7254 sp<EffectChain> chain = mChain.promote(); 7255 if (chain != 0 && chain->activeTrackCnt() != 0) { 7256 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7257 int16_t *in = mConfig.inputCfg.buffer.s16; 7258 int16_t *out = mConfig.outputCfg.buffer.s16; 7259 for (size_t i = 0; i < frameCnt; i++) { 7260 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7261 } 7262 } 7263 } 7264} 7265 7266void AudioFlinger::EffectModule::reset_l() 7267{ 7268 if (mEffectInterface == NULL) { 7269 return; 7270 } 7271 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7272} 7273 7274status_t AudioFlinger::EffectModule::configure() 7275{ 7276 uint32_t channels; 7277 if (mEffectInterface == NULL) { 7278 return NO_INIT; 7279 } 7280 7281 sp<ThreadBase> thread = mThread.promote(); 7282 if (thread == 0) { 7283 return DEAD_OBJECT; 7284 } 7285 7286 // TODO: handle configuration of effects replacing track process 7287 if (thread->channelCount() == 1) { 7288 channels = AUDIO_CHANNEL_OUT_MONO; 7289 } else { 7290 channels = AUDIO_CHANNEL_OUT_STEREO; 7291 } 7292 7293 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7294 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7295 } else { 7296 mConfig.inputCfg.channels = channels; 7297 } 7298 mConfig.outputCfg.channels = channels; 7299 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7300 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7301 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7302 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7303 mConfig.inputCfg.bufferProvider.cookie = NULL; 7304 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7305 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7306 mConfig.outputCfg.bufferProvider.cookie = NULL; 7307 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7308 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7309 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7310 // Insert effect: 7311 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7312 // always overwrites output buffer: input buffer == output buffer 7313 // - in other sessions: 7314 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7315 // other effect: overwrites output buffer: input buffer == output buffer 7316 // Auxiliary effect: 7317 // accumulates in output buffer: input buffer != output buffer 7318 // Therefore: accumulate <=> input buffer != output buffer 7319 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7320 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7321 } else { 7322 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7323 } 7324 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7325 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7326 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7327 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7328 7329 ALOGV("configure() %p thread %p buffer %p framecount %d", 7330 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7331 7332 status_t cmdStatus; 7333 uint32_t size = sizeof(int); 7334 status_t status = (*mEffectInterface)->command(mEffectInterface, 7335 EFFECT_CMD_SET_CONFIG, 7336 sizeof(effect_config_t), 7337 &mConfig, 7338 &size, 7339 &cmdStatus); 7340 if (status == 0) { 7341 status = cmdStatus; 7342 } 7343 7344 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7345 (1000 * mConfig.outputCfg.buffer.frameCount); 7346 7347 return status; 7348} 7349 7350status_t AudioFlinger::EffectModule::init() 7351{ 7352 Mutex::Autolock _l(mLock); 7353 if (mEffectInterface == NULL) { 7354 return NO_INIT; 7355 } 7356 status_t cmdStatus; 7357 uint32_t size = sizeof(status_t); 7358 status_t status = (*mEffectInterface)->command(mEffectInterface, 7359 EFFECT_CMD_INIT, 7360 0, 7361 NULL, 7362 &size, 7363 &cmdStatus); 7364 if (status == 0) { 7365 status = cmdStatus; 7366 } 7367 return status; 7368} 7369 7370status_t AudioFlinger::EffectModule::start() 7371{ 7372 Mutex::Autolock _l(mLock); 7373 return start_l(); 7374} 7375 7376status_t AudioFlinger::EffectModule::start_l() 7377{ 7378 if (mEffectInterface == NULL) { 7379 return NO_INIT; 7380 } 7381 status_t cmdStatus; 7382 uint32_t size = sizeof(status_t); 7383 status_t status = (*mEffectInterface)->command(mEffectInterface, 7384 EFFECT_CMD_ENABLE, 7385 0, 7386 NULL, 7387 &size, 7388 &cmdStatus); 7389 if (status == 0) { 7390 status = cmdStatus; 7391 } 7392 if (status == 0 && 7393 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7394 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7395 sp<ThreadBase> thread = mThread.promote(); 7396 if (thread != 0) { 7397 audio_stream_t *stream = thread->stream(); 7398 if (stream != NULL) { 7399 stream->add_audio_effect(stream, mEffectInterface); 7400 } 7401 } 7402 } 7403 return status; 7404} 7405 7406status_t AudioFlinger::EffectModule::stop() 7407{ 7408 Mutex::Autolock _l(mLock); 7409 return stop_l(); 7410} 7411 7412status_t AudioFlinger::EffectModule::stop_l() 7413{ 7414 if (mEffectInterface == NULL) { 7415 return NO_INIT; 7416 } 7417 status_t cmdStatus; 7418 uint32_t size = sizeof(status_t); 7419 status_t status = (*mEffectInterface)->command(mEffectInterface, 7420 EFFECT_CMD_DISABLE, 7421 0, 7422 NULL, 7423 &size, 7424 &cmdStatus); 7425 if (status == 0) { 7426 status = cmdStatus; 7427 } 7428 if (status == 0 && 7429 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7430 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7431 sp<ThreadBase> thread = mThread.promote(); 7432 if (thread != 0) { 7433 audio_stream_t *stream = thread->stream(); 7434 if (stream != NULL) { 7435 stream->remove_audio_effect(stream, mEffectInterface); 7436 } 7437 } 7438 } 7439 return status; 7440} 7441 7442status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7443 uint32_t cmdSize, 7444 void *pCmdData, 7445 uint32_t *replySize, 7446 void *pReplyData) 7447{ 7448 Mutex::Autolock _l(mLock); 7449// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7450 7451 if (mState == DESTROYED || mEffectInterface == NULL) { 7452 return NO_INIT; 7453 } 7454 status_t status = (*mEffectInterface)->command(mEffectInterface, 7455 cmdCode, 7456 cmdSize, 7457 pCmdData, 7458 replySize, 7459 pReplyData); 7460 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7461 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7462 for (size_t i = 1; i < mHandles.size(); i++) { 7463 sp<EffectHandle> h = mHandles[i].promote(); 7464 if (h != 0) { 7465 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7466 } 7467 } 7468 } 7469 return status; 7470} 7471 7472status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7473{ 7474 7475 Mutex::Autolock _l(mLock); 7476 ALOGV("setEnabled %p enabled %d", this, enabled); 7477 7478 if (enabled != isEnabled()) { 7479 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7480 if (enabled && status != NO_ERROR) { 7481 return status; 7482 } 7483 7484 switch (mState) { 7485 // going from disabled to enabled 7486 case IDLE: 7487 mState = STARTING; 7488 break; 7489 case STOPPED: 7490 mState = RESTART; 7491 break; 7492 case STOPPING: 7493 mState = ACTIVE; 7494 break; 7495 7496 // going from enabled to disabled 7497 case RESTART: 7498 mState = STOPPED; 7499 break; 7500 case STARTING: 7501 mState = IDLE; 7502 break; 7503 case ACTIVE: 7504 mState = STOPPING; 7505 break; 7506 case DESTROYED: 7507 return NO_ERROR; // simply ignore as we are being destroyed 7508 } 7509 for (size_t i = 1; i < mHandles.size(); i++) { 7510 sp<EffectHandle> h = mHandles[i].promote(); 7511 if (h != 0) { 7512 h->setEnabled(enabled); 7513 } 7514 } 7515 } 7516 return NO_ERROR; 7517} 7518 7519bool AudioFlinger::EffectModule::isEnabled() const 7520{ 7521 switch (mState) { 7522 case RESTART: 7523 case STARTING: 7524 case ACTIVE: 7525 return true; 7526 case IDLE: 7527 case STOPPING: 7528 case STOPPED: 7529 case DESTROYED: 7530 default: 7531 return false; 7532 } 7533} 7534 7535bool AudioFlinger::EffectModule::isProcessEnabled() const 7536{ 7537 switch (mState) { 7538 case RESTART: 7539 case ACTIVE: 7540 case STOPPING: 7541 case STOPPED: 7542 return true; 7543 case IDLE: 7544 case STARTING: 7545 case DESTROYED: 7546 default: 7547 return false; 7548 } 7549} 7550 7551status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7552{ 7553 Mutex::Autolock _l(mLock); 7554 status_t status = NO_ERROR; 7555 7556 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7557 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7558 if (isProcessEnabled() && 7559 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7560 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7561 status_t cmdStatus; 7562 uint32_t volume[2]; 7563 uint32_t *pVolume = NULL; 7564 uint32_t size = sizeof(volume); 7565 volume[0] = *left; 7566 volume[1] = *right; 7567 if (controller) { 7568 pVolume = volume; 7569 } 7570 status = (*mEffectInterface)->command(mEffectInterface, 7571 EFFECT_CMD_SET_VOLUME, 7572 size, 7573 volume, 7574 &size, 7575 pVolume); 7576 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7577 *left = volume[0]; 7578 *right = volume[1]; 7579 } 7580 } 7581 return status; 7582} 7583 7584status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7585{ 7586 Mutex::Autolock _l(mLock); 7587 status_t status = NO_ERROR; 7588 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7589 // audio pre processing modules on RecordThread can receive both output and 7590 // input device indication in the same call 7591 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7592 if (dev) { 7593 status_t cmdStatus; 7594 uint32_t size = sizeof(status_t); 7595 7596 status = (*mEffectInterface)->command(mEffectInterface, 7597 EFFECT_CMD_SET_DEVICE, 7598 sizeof(uint32_t), 7599 &dev, 7600 &size, 7601 &cmdStatus); 7602 if (status == NO_ERROR) { 7603 status = cmdStatus; 7604 } 7605 } 7606 dev = device & AUDIO_DEVICE_IN_ALL; 7607 if (dev) { 7608 status_t cmdStatus; 7609 uint32_t size = sizeof(status_t); 7610 7611 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7612 EFFECT_CMD_SET_INPUT_DEVICE, 7613 sizeof(uint32_t), 7614 &dev, 7615 &size, 7616 &cmdStatus); 7617 if (status2 == NO_ERROR) { 7618 status2 = cmdStatus; 7619 } 7620 if (status == NO_ERROR) { 7621 status = status2; 7622 } 7623 } 7624 } 7625 return status; 7626} 7627 7628status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7629{ 7630 Mutex::Autolock _l(mLock); 7631 status_t status = NO_ERROR; 7632 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7633 status_t cmdStatus; 7634 uint32_t size = sizeof(status_t); 7635 status = (*mEffectInterface)->command(mEffectInterface, 7636 EFFECT_CMD_SET_AUDIO_MODE, 7637 sizeof(audio_mode_t), 7638 &mode, 7639 &size, 7640 &cmdStatus); 7641 if (status == NO_ERROR) { 7642 status = cmdStatus; 7643 } 7644 } 7645 return status; 7646} 7647 7648void AudioFlinger::EffectModule::setSuspended(bool suspended) 7649{ 7650 Mutex::Autolock _l(mLock); 7651 mSuspended = suspended; 7652} 7653 7654bool AudioFlinger::EffectModule::suspended() const 7655{ 7656 Mutex::Autolock _l(mLock); 7657 return mSuspended; 7658} 7659 7660status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7661{ 7662 const size_t SIZE = 256; 7663 char buffer[SIZE]; 7664 String8 result; 7665 7666 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7667 result.append(buffer); 7668 7669 bool locked = tryLock(mLock); 7670 // failed to lock - AudioFlinger is probably deadlocked 7671 if (!locked) { 7672 result.append("\t\tCould not lock Fx mutex:\n"); 7673 } 7674 7675 result.append("\t\tSession Status State Engine:\n"); 7676 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7677 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7678 result.append(buffer); 7679 7680 result.append("\t\tDescriptor:\n"); 7681 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7682 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7683 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7684 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7685 result.append(buffer); 7686 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7687 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7688 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7689 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7690 result.append(buffer); 7691 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7692 mDescriptor.apiVersion, 7693 mDescriptor.flags); 7694 result.append(buffer); 7695 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7696 mDescriptor.name); 7697 result.append(buffer); 7698 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7699 mDescriptor.implementor); 7700 result.append(buffer); 7701 7702 result.append("\t\t- Input configuration:\n"); 7703 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7704 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7705 (uint32_t)mConfig.inputCfg.buffer.raw, 7706 mConfig.inputCfg.buffer.frameCount, 7707 mConfig.inputCfg.samplingRate, 7708 mConfig.inputCfg.channels, 7709 mConfig.inputCfg.format); 7710 result.append(buffer); 7711 7712 result.append("\t\t- Output configuration:\n"); 7713 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7714 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7715 (uint32_t)mConfig.outputCfg.buffer.raw, 7716 mConfig.outputCfg.buffer.frameCount, 7717 mConfig.outputCfg.samplingRate, 7718 mConfig.outputCfg.channels, 7719 mConfig.outputCfg.format); 7720 result.append(buffer); 7721 7722 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7723 result.append(buffer); 7724 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7725 for (size_t i = 0; i < mHandles.size(); ++i) { 7726 sp<EffectHandle> handle = mHandles[i].promote(); 7727 if (handle != 0) { 7728 handle->dump(buffer, SIZE); 7729 result.append(buffer); 7730 } 7731 } 7732 7733 result.append("\n"); 7734 7735 write(fd, result.string(), result.length()); 7736 7737 if (locked) { 7738 mLock.unlock(); 7739 } 7740 7741 return NO_ERROR; 7742} 7743 7744// ---------------------------------------------------------------------------- 7745// EffectHandle implementation 7746// ---------------------------------------------------------------------------- 7747 7748#undef LOG_TAG 7749#define LOG_TAG "AudioFlinger::EffectHandle" 7750 7751AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7752 const sp<AudioFlinger::Client>& client, 7753 const sp<IEffectClient>& effectClient, 7754 int32_t priority) 7755 : BnEffect(), 7756 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7757 mPriority(priority), mHasControl(false), mEnabled(false) 7758{ 7759 ALOGV("constructor %p", this); 7760 7761 if (client == 0) { 7762 return; 7763 } 7764 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7765 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7766 if (mCblkMemory != 0) { 7767 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7768 7769 if (mCblk != NULL) { 7770 new(mCblk) effect_param_cblk_t(); 7771 mBuffer = (uint8_t *)mCblk + bufOffset; 7772 } 7773 } else { 7774 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7775 return; 7776 } 7777} 7778 7779AudioFlinger::EffectHandle::~EffectHandle() 7780{ 7781 ALOGV("Destructor %p", this); 7782 disconnect(false); 7783 ALOGV("Destructor DONE %p", this); 7784} 7785 7786status_t AudioFlinger::EffectHandle::enable() 7787{ 7788 ALOGV("enable %p", this); 7789 if (!mHasControl) return INVALID_OPERATION; 7790 if (mEffect == 0) return DEAD_OBJECT; 7791 7792 if (mEnabled) { 7793 return NO_ERROR; 7794 } 7795 7796 mEnabled = true; 7797 7798 sp<ThreadBase> thread = mEffect->thread().promote(); 7799 if (thread != 0) { 7800 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7801 } 7802 7803 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7804 if (mEffect->suspended()) { 7805 return NO_ERROR; 7806 } 7807 7808 status_t status = mEffect->setEnabled(true); 7809 if (status != NO_ERROR) { 7810 if (thread != 0) { 7811 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7812 } 7813 mEnabled = false; 7814 } 7815 return status; 7816} 7817 7818status_t AudioFlinger::EffectHandle::disable() 7819{ 7820 ALOGV("disable %p", this); 7821 if (!mHasControl) return INVALID_OPERATION; 7822 if (mEffect == 0) return DEAD_OBJECT; 7823 7824 if (!mEnabled) { 7825 return NO_ERROR; 7826 } 7827 mEnabled = false; 7828 7829 if (mEffect->suspended()) { 7830 return NO_ERROR; 7831 } 7832 7833 status_t status = mEffect->setEnabled(false); 7834 7835 sp<ThreadBase> thread = mEffect->thread().promote(); 7836 if (thread != 0) { 7837 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7838 } 7839 7840 return status; 7841} 7842 7843void AudioFlinger::EffectHandle::disconnect() 7844{ 7845 disconnect(true); 7846} 7847 7848void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7849{ 7850 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7851 if (mEffect == 0) { 7852 return; 7853 } 7854 mEffect->disconnect(this, unpinIfLast); 7855 7856 if (mHasControl && mEnabled) { 7857 sp<ThreadBase> thread = mEffect->thread().promote(); 7858 if (thread != 0) { 7859 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7860 } 7861 } 7862 7863 // release sp on module => module destructor can be called now 7864 mEffect.clear(); 7865 if (mClient != 0) { 7866 if (mCblk != NULL) { 7867 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7868 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7869 } 7870 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7871 // Client destructor must run with AudioFlinger mutex locked 7872 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7873 mClient.clear(); 7874 } 7875} 7876 7877status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7878 uint32_t cmdSize, 7879 void *pCmdData, 7880 uint32_t *replySize, 7881 void *pReplyData) 7882{ 7883// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7884// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7885 7886 // only get parameter command is permitted for applications not controlling the effect 7887 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7888 return INVALID_OPERATION; 7889 } 7890 if (mEffect == 0) return DEAD_OBJECT; 7891 if (mClient == 0) return INVALID_OPERATION; 7892 7893 // handle commands that are not forwarded transparently to effect engine 7894 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7895 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7896 // no risk to block the whole media server process or mixer threads is we are stuck here 7897 Mutex::Autolock _l(mCblk->lock); 7898 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7899 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7900 mCblk->serverIndex = 0; 7901 mCblk->clientIndex = 0; 7902 return BAD_VALUE; 7903 } 7904 status_t status = NO_ERROR; 7905 while (mCblk->serverIndex < mCblk->clientIndex) { 7906 int reply; 7907 uint32_t rsize = sizeof(int); 7908 int *p = (int *)(mBuffer + mCblk->serverIndex); 7909 int size = *p++; 7910 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7911 ALOGW("command(): invalid parameter block size"); 7912 break; 7913 } 7914 effect_param_t *param = (effect_param_t *)p; 7915 if (param->psize == 0 || param->vsize == 0) { 7916 ALOGW("command(): null parameter or value size"); 7917 mCblk->serverIndex += size; 7918 continue; 7919 } 7920 uint32_t psize = sizeof(effect_param_t) + 7921 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7922 param->vsize; 7923 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7924 psize, 7925 p, 7926 &rsize, 7927 &reply); 7928 // stop at first error encountered 7929 if (ret != NO_ERROR) { 7930 status = ret; 7931 *(int *)pReplyData = reply; 7932 break; 7933 } else if (reply != NO_ERROR) { 7934 *(int *)pReplyData = reply; 7935 break; 7936 } 7937 mCblk->serverIndex += size; 7938 } 7939 mCblk->serverIndex = 0; 7940 mCblk->clientIndex = 0; 7941 return status; 7942 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7943 *(int *)pReplyData = NO_ERROR; 7944 return enable(); 7945 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7946 *(int *)pReplyData = NO_ERROR; 7947 return disable(); 7948 } 7949 7950 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7951} 7952 7953void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7954{ 7955 ALOGV("setControl %p control %d", this, hasControl); 7956 7957 mHasControl = hasControl; 7958 mEnabled = enabled; 7959 7960 if (signal && mEffectClient != 0) { 7961 mEffectClient->controlStatusChanged(hasControl); 7962 } 7963} 7964 7965void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7966 uint32_t cmdSize, 7967 void *pCmdData, 7968 uint32_t replySize, 7969 void *pReplyData) 7970{ 7971 if (mEffectClient != 0) { 7972 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7973 } 7974} 7975 7976 7977 7978void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7979{ 7980 if (mEffectClient != 0) { 7981 mEffectClient->enableStatusChanged(enabled); 7982 } 7983} 7984 7985status_t AudioFlinger::EffectHandle::onTransact( 7986 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7987{ 7988 return BnEffect::onTransact(code, data, reply, flags); 7989} 7990 7991 7992void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7993{ 7994 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7995 7996 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7997 (mClient == 0) ? getpid_cached : mClient->pid(), 7998 mPriority, 7999 mHasControl, 8000 !locked, 8001 mCblk ? mCblk->clientIndex : 0, 8002 mCblk ? mCblk->serverIndex : 0 8003 ); 8004 8005 if (locked) { 8006 mCblk->lock.unlock(); 8007 } 8008} 8009 8010#undef LOG_TAG 8011#define LOG_TAG "AudioFlinger::EffectChain" 8012 8013AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8014 int sessionId) 8015 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8016 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8017 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8018{ 8019 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8020 if (thread == NULL) { 8021 return; 8022 } 8023 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8024 thread->frameCount(); 8025} 8026 8027AudioFlinger::EffectChain::~EffectChain() 8028{ 8029 if (mOwnInBuffer) { 8030 delete mInBuffer; 8031 } 8032 8033} 8034 8035// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8036sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8037{ 8038 size_t size = mEffects.size(); 8039 8040 for (size_t i = 0; i < size; i++) { 8041 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8042 return mEffects[i]; 8043 } 8044 } 8045 return 0; 8046} 8047 8048// getEffectFromId_l() must be called with ThreadBase::mLock held 8049sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8050{ 8051 size_t size = mEffects.size(); 8052 8053 for (size_t i = 0; i < size; i++) { 8054 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8055 if (id == 0 || mEffects[i]->id() == id) { 8056 return mEffects[i]; 8057 } 8058 } 8059 return 0; 8060} 8061 8062// getEffectFromType_l() must be called with ThreadBase::mLock held 8063sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8064 const effect_uuid_t *type) 8065{ 8066 size_t size = mEffects.size(); 8067 8068 for (size_t i = 0; i < size; i++) { 8069 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8070 return mEffects[i]; 8071 } 8072 } 8073 return 0; 8074} 8075 8076// Must be called with EffectChain::mLock locked 8077void AudioFlinger::EffectChain::process_l() 8078{ 8079 sp<ThreadBase> thread = mThread.promote(); 8080 if (thread == 0) { 8081 ALOGW("process_l(): cannot promote mixer thread"); 8082 return; 8083 } 8084 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8085 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8086 // always process effects unless no more tracks are on the session and the effect tail 8087 // has been rendered 8088 bool doProcess = true; 8089 if (!isGlobalSession) { 8090 bool tracksOnSession = (trackCnt() != 0); 8091 8092 if (!tracksOnSession && mTailBufferCount == 0) { 8093 doProcess = false; 8094 } 8095 8096 if (activeTrackCnt() == 0) { 8097 // if no track is active and the effect tail has not been rendered, 8098 // the input buffer must be cleared here as the mixer process will not do it 8099 if (tracksOnSession || mTailBufferCount > 0) { 8100 size_t numSamples = thread->frameCount() * thread->channelCount(); 8101 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8102 if (mTailBufferCount > 0) { 8103 mTailBufferCount--; 8104 } 8105 } 8106 } 8107 } 8108 8109 size_t size = mEffects.size(); 8110 if (doProcess) { 8111 for (size_t i = 0; i < size; i++) { 8112 mEffects[i]->process(); 8113 } 8114 } 8115 for (size_t i = 0; i < size; i++) { 8116 mEffects[i]->updateState(); 8117 } 8118} 8119 8120// addEffect_l() must be called with PlaybackThread::mLock held 8121status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8122{ 8123 effect_descriptor_t desc = effect->desc(); 8124 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8125 8126 Mutex::Autolock _l(mLock); 8127 effect->setChain(this); 8128 sp<ThreadBase> thread = mThread.promote(); 8129 if (thread == 0) { 8130 return NO_INIT; 8131 } 8132 effect->setThread(thread); 8133 8134 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8135 // Auxiliary effects are inserted at the beginning of mEffects vector as 8136 // they are processed first and accumulated in chain input buffer 8137 mEffects.insertAt(effect, 0); 8138 8139 // the input buffer for auxiliary effect contains mono samples in 8140 // 32 bit format. This is to avoid saturation in AudoMixer 8141 // accumulation stage. Saturation is done in EffectModule::process() before 8142 // calling the process in effect engine 8143 size_t numSamples = thread->frameCount(); 8144 int32_t *buffer = new int32_t[numSamples]; 8145 memset(buffer, 0, numSamples * sizeof(int32_t)); 8146 effect->setInBuffer((int16_t *)buffer); 8147 // auxiliary effects output samples to chain input buffer for further processing 8148 // by insert effects 8149 effect->setOutBuffer(mInBuffer); 8150 } else { 8151 // Insert effects are inserted at the end of mEffects vector as they are processed 8152 // after track and auxiliary effects. 8153 // Insert effect order as a function of indicated preference: 8154 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8155 // another effect is present 8156 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8157 // last effect claiming first position 8158 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8159 // first effect claiming last position 8160 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8161 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8162 // already present 8163 8164 size_t size = mEffects.size(); 8165 size_t idx_insert = size; 8166 ssize_t idx_insert_first = -1; 8167 ssize_t idx_insert_last = -1; 8168 8169 for (size_t i = 0; i < size; i++) { 8170 effect_descriptor_t d = mEffects[i]->desc(); 8171 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8172 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8173 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8174 // check invalid effect chaining combinations 8175 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8176 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8177 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8178 return INVALID_OPERATION; 8179 } 8180 // remember position of first insert effect and by default 8181 // select this as insert position for new effect 8182 if (idx_insert == size) { 8183 idx_insert = i; 8184 } 8185 // remember position of last insert effect claiming 8186 // first position 8187 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8188 idx_insert_first = i; 8189 } 8190 // remember position of first insert effect claiming 8191 // last position 8192 if (iPref == EFFECT_FLAG_INSERT_LAST && 8193 idx_insert_last == -1) { 8194 idx_insert_last = i; 8195 } 8196 } 8197 } 8198 8199 // modify idx_insert from first position if needed 8200 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8201 if (idx_insert_last != -1) { 8202 idx_insert = idx_insert_last; 8203 } else { 8204 idx_insert = size; 8205 } 8206 } else { 8207 if (idx_insert_first != -1) { 8208 idx_insert = idx_insert_first + 1; 8209 } 8210 } 8211 8212 // always read samples from chain input buffer 8213 effect->setInBuffer(mInBuffer); 8214 8215 // if last effect in the chain, output samples to chain 8216 // output buffer, otherwise to chain input buffer 8217 if (idx_insert == size) { 8218 if (idx_insert != 0) { 8219 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8220 mEffects[idx_insert-1]->configure(); 8221 } 8222 effect->setOutBuffer(mOutBuffer); 8223 } else { 8224 effect->setOutBuffer(mInBuffer); 8225 } 8226 mEffects.insertAt(effect, idx_insert); 8227 8228 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8229 } 8230 effect->configure(); 8231 return NO_ERROR; 8232} 8233 8234// removeEffect_l() must be called with PlaybackThread::mLock held 8235size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8236{ 8237 Mutex::Autolock _l(mLock); 8238 size_t size = mEffects.size(); 8239 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8240 8241 for (size_t i = 0; i < size; i++) { 8242 if (effect == mEffects[i]) { 8243 // calling stop here will remove pre-processing effect from the audio HAL. 8244 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8245 // the middle of a read from audio HAL 8246 if (mEffects[i]->state() == EffectModule::ACTIVE || 8247 mEffects[i]->state() == EffectModule::STOPPING) { 8248 mEffects[i]->stop(); 8249 } 8250 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8251 delete[] effect->inBuffer(); 8252 } else { 8253 if (i == size - 1 && i != 0) { 8254 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8255 mEffects[i - 1]->configure(); 8256 } 8257 } 8258 mEffects.removeAt(i); 8259 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8260 break; 8261 } 8262 } 8263 8264 return mEffects.size(); 8265} 8266 8267// setDevice_l() must be called with PlaybackThread::mLock held 8268void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8269{ 8270 size_t size = mEffects.size(); 8271 for (size_t i = 0; i < size; i++) { 8272 mEffects[i]->setDevice(device); 8273 } 8274} 8275 8276// setMode_l() must be called with PlaybackThread::mLock held 8277void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8278{ 8279 size_t size = mEffects.size(); 8280 for (size_t i = 0; i < size; i++) { 8281 mEffects[i]->setMode(mode); 8282 } 8283} 8284 8285// setVolume_l() must be called with PlaybackThread::mLock held 8286bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8287{ 8288 uint32_t newLeft = *left; 8289 uint32_t newRight = *right; 8290 bool hasControl = false; 8291 int ctrlIdx = -1; 8292 size_t size = mEffects.size(); 8293 8294 // first update volume controller 8295 for (size_t i = size; i > 0; i--) { 8296 if (mEffects[i - 1]->isProcessEnabled() && 8297 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8298 ctrlIdx = i - 1; 8299 hasControl = true; 8300 break; 8301 } 8302 } 8303 8304 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8305 if (hasControl) { 8306 *left = mNewLeftVolume; 8307 *right = mNewRightVolume; 8308 } 8309 return hasControl; 8310 } 8311 8312 mVolumeCtrlIdx = ctrlIdx; 8313 mLeftVolume = newLeft; 8314 mRightVolume = newRight; 8315 8316 // second get volume update from volume controller 8317 if (ctrlIdx >= 0) { 8318 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8319 mNewLeftVolume = newLeft; 8320 mNewRightVolume = newRight; 8321 } 8322 // then indicate volume to all other effects in chain. 8323 // Pass altered volume to effects before volume controller 8324 // and requested volume to effects after controller 8325 uint32_t lVol = newLeft; 8326 uint32_t rVol = newRight; 8327 8328 for (size_t i = 0; i < size; i++) { 8329 if ((int)i == ctrlIdx) continue; 8330 // this also works for ctrlIdx == -1 when there is no volume controller 8331 if ((int)i > ctrlIdx) { 8332 lVol = *left; 8333 rVol = *right; 8334 } 8335 mEffects[i]->setVolume(&lVol, &rVol, false); 8336 } 8337 *left = newLeft; 8338 *right = newRight; 8339 8340 return hasControl; 8341} 8342 8343status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8344{ 8345 const size_t SIZE = 256; 8346 char buffer[SIZE]; 8347 String8 result; 8348 8349 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8350 result.append(buffer); 8351 8352 bool locked = tryLock(mLock); 8353 // failed to lock - AudioFlinger is probably deadlocked 8354 if (!locked) { 8355 result.append("\tCould not lock mutex:\n"); 8356 } 8357 8358 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8359 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8360 mEffects.size(), 8361 (uint32_t)mInBuffer, 8362 (uint32_t)mOutBuffer, 8363 mActiveTrackCnt); 8364 result.append(buffer); 8365 write(fd, result.string(), result.size()); 8366 8367 for (size_t i = 0; i < mEffects.size(); ++i) { 8368 sp<EffectModule> effect = mEffects[i]; 8369 if (effect != 0) { 8370 effect->dump(fd, args); 8371 } 8372 } 8373 8374 if (locked) { 8375 mLock.unlock(); 8376 } 8377 8378 return NO_ERROR; 8379} 8380 8381// must be called with ThreadBase::mLock held 8382void AudioFlinger::EffectChain::setEffectSuspended_l( 8383 const effect_uuid_t *type, bool suspend) 8384{ 8385 sp<SuspendedEffectDesc> desc; 8386 // use effect type UUID timelow as key as there is no real risk of identical 8387 // timeLow fields among effect type UUIDs. 8388 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8389 if (suspend) { 8390 if (index >= 0) { 8391 desc = mSuspendedEffects.valueAt(index); 8392 } else { 8393 desc = new SuspendedEffectDesc(); 8394 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8395 mSuspendedEffects.add(type->timeLow, desc); 8396 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8397 } 8398 if (desc->mRefCount++ == 0) { 8399 sp<EffectModule> effect = getEffectIfEnabled(type); 8400 if (effect != 0) { 8401 desc->mEffect = effect; 8402 effect->setSuspended(true); 8403 effect->setEnabled(false); 8404 } 8405 } 8406 } else { 8407 if (index < 0) { 8408 return; 8409 } 8410 desc = mSuspendedEffects.valueAt(index); 8411 if (desc->mRefCount <= 0) { 8412 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8413 desc->mRefCount = 1; 8414 } 8415 if (--desc->mRefCount == 0) { 8416 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8417 if (desc->mEffect != 0) { 8418 sp<EffectModule> effect = desc->mEffect.promote(); 8419 if (effect != 0) { 8420 effect->setSuspended(false); 8421 sp<EffectHandle> handle = effect->controlHandle(); 8422 if (handle != 0) { 8423 effect->setEnabled(handle->enabled()); 8424 } 8425 } 8426 desc->mEffect.clear(); 8427 } 8428 mSuspendedEffects.removeItemsAt(index); 8429 } 8430 } 8431} 8432 8433// must be called with ThreadBase::mLock held 8434void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8435{ 8436 sp<SuspendedEffectDesc> desc; 8437 8438 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8439 if (suspend) { 8440 if (index >= 0) { 8441 desc = mSuspendedEffects.valueAt(index); 8442 } else { 8443 desc = new SuspendedEffectDesc(); 8444 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8445 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8446 } 8447 if (desc->mRefCount++ == 0) { 8448 Vector< sp<EffectModule> > effects; 8449 getSuspendEligibleEffects(effects); 8450 for (size_t i = 0; i < effects.size(); i++) { 8451 setEffectSuspended_l(&effects[i]->desc().type, true); 8452 } 8453 } 8454 } else { 8455 if (index < 0) { 8456 return; 8457 } 8458 desc = mSuspendedEffects.valueAt(index); 8459 if (desc->mRefCount <= 0) { 8460 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8461 desc->mRefCount = 1; 8462 } 8463 if (--desc->mRefCount == 0) { 8464 Vector<const effect_uuid_t *> types; 8465 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8466 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8467 continue; 8468 } 8469 types.add(&mSuspendedEffects.valueAt(i)->mType); 8470 } 8471 for (size_t i = 0; i < types.size(); i++) { 8472 setEffectSuspended_l(types[i], false); 8473 } 8474 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8475 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8476 } 8477 } 8478} 8479 8480 8481// The volume effect is used for automated tests only 8482#ifndef OPENSL_ES_H_ 8483static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8484 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8485const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8486#endif //OPENSL_ES_H_ 8487 8488bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8489{ 8490 // auxiliary effects and visualizer are never suspended on output mix 8491 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8492 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8493 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8494 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8495 return false; 8496 } 8497 return true; 8498} 8499 8500void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8501{ 8502 effects.clear(); 8503 for (size_t i = 0; i < mEffects.size(); i++) { 8504 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8505 effects.add(mEffects[i]); 8506 } 8507 } 8508} 8509 8510sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8511 const effect_uuid_t *type) 8512{ 8513 sp<EffectModule> effect = getEffectFromType_l(type); 8514 return effect != 0 && effect->isEnabled() ? effect : 0; 8515} 8516 8517void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8518 bool enabled) 8519{ 8520 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8521 if (enabled) { 8522 if (index < 0) { 8523 // if the effect is not suspend check if all effects are suspended 8524 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8525 if (index < 0) { 8526 return; 8527 } 8528 if (!isEffectEligibleForSuspend(effect->desc())) { 8529 return; 8530 } 8531 setEffectSuspended_l(&effect->desc().type, enabled); 8532 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8533 if (index < 0) { 8534 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8535 return; 8536 } 8537 } 8538 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8539 effect->desc().type.timeLow); 8540 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8541 // if effect is requested to suspended but was not yet enabled, supend it now. 8542 if (desc->mEffect == 0) { 8543 desc->mEffect = effect; 8544 effect->setEnabled(false); 8545 effect->setSuspended(true); 8546 } 8547 } else { 8548 if (index < 0) { 8549 return; 8550 } 8551 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8552 effect->desc().type.timeLow); 8553 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8554 desc->mEffect.clear(); 8555 effect->setSuspended(false); 8556 } 8557} 8558 8559#undef LOG_TAG 8560#define LOG_TAG "AudioFlinger" 8561 8562// ---------------------------------------------------------------------------- 8563 8564status_t AudioFlinger::onTransact( 8565 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8566{ 8567 return BnAudioFlinger::onTransact(code, data, reply, flags); 8568} 8569 8570}; // namespace android 8571