AudioFlinger.cpp revision a4c5a550e2a3bc237179b8684e51718e05894492
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 78namespace android { 79 80static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 81static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 82 83static const float MAX_GAIN = 4096.0f; 84static const uint32_t MAX_GAIN_INT = 0x1000; 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95static const int kDumpLockRetries = 50; 96static const int kDumpLockSleepUs = 20000; 97 98// don't warn about blocked writes or record buffer overflows more often than this 99static const nsecs_t kWarningThrottleNs = seconds(5); 100 101// RecordThread loop sleep time upon application overrun or audio HAL read error 102static const int kRecordThreadSleepUs = 5000; 103 104// maximum time to wait for setParameters to complete 105static const nsecs_t kSetParametersTimeoutNs = seconds(2); 106 107// minimum sleep time for the mixer thread loop when tracks are active but in underrun 108static const uint32_t kMinThreadSleepTimeUs = 5000; 109// maximum divider applied to the active sleep time in the mixer thread loop 110static const uint32_t kMaxThreadSleepTimeShift = 2; 111 112nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 113 114// ---------------------------------------------------------------------------- 115 116#ifdef ADD_BATTERY_DATA 117// To collect the amplifier usage 118static void addBatteryData(uint32_t params) { 119 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 120 if (service == NULL) { 121 // it already logged 122 return; 123 } 124 125 service->addBatteryData(params); 126} 127#endif 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152// ---------------------------------------------------------------------------- 153 154AudioFlinger::AudioFlinger() 155 : BnAudioFlinger(), 156 mPrimaryHardwareDev(NULL), 157 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 158 mMasterVolume(1.0f), 159 mMasterVolumeSupportLvl(MVS_NONE), 160 mMasterMute(false), 161 mNextUniqueId(1), 162 mMode(AUDIO_MODE_INVALID), 163 mBtNrecIsOff(false) 164{ 165} 166 167void AudioFlinger::onFirstRef() 168{ 169 int rc = 0; 170 171 Mutex::Autolock _l(mLock); 172 173 /* TODO: move all this work into an Init() function */ 174 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 175 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 176 uint32_t int_val; 177 if (1 == sscanf(val_str, "%u", &int_val)) { 178 mStandbyTimeInNsecs = milliseconds(int_val); 179 ALOGI("Using %u mSec as standby time.", int_val); 180 } else { 181 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 182 ALOGI("Using default %u mSec as standby time.", 183 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 184 } 185 } 186 187 mMode = AUDIO_MODE_NORMAL; 188 mMasterVolumeSW = 1.0; 189 mMasterVolume = 1.0; 190 mHardwareStatus = AUDIO_HW_IDLE; 191} 192 193AudioFlinger::~AudioFlinger() 194{ 195 196 while (!mRecordThreads.isEmpty()) { 197 // closeInput() will remove first entry from mRecordThreads 198 closeInput(mRecordThreads.keyAt(0)); 199 } 200 while (!mPlaybackThreads.isEmpty()) { 201 // closeOutput() will remove first entry from mPlaybackThreads 202 closeOutput(mPlaybackThreads.keyAt(0)); 203 } 204 205 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 206 // no mHardwareLock needed, as there are no other references to this 207 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 208 delete mAudioHwDevs.valueAt(i); 209 } 210} 211 212static const char * const audio_interfaces[] = { 213 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 214 AUDIO_HARDWARE_MODULE_ID_A2DP, 215 AUDIO_HARDWARE_MODULE_ID_USB, 216}; 217#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 218 219audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 220{ 221 // if module is 0, the request comes from an old policy manager and we should load 222 // well known modules 223 if (module == 0) { 224 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 225 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 226 loadHwModule_l(audio_interfaces[i]); 227 } 228 } else { 229 // check a match for the requested module handle 230 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 231 if (audioHwdevice != NULL) { 232 return audioHwdevice->hwDevice(); 233 } 234 } 235 // then try to find a module supporting the requested device. 236 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 237 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 238 if ((dev->get_supported_devices(dev) & devices) == devices) 239 return dev; 240 } 241 242 return NULL; 243} 244 245status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 246{ 247 const size_t SIZE = 256; 248 char buffer[SIZE]; 249 String8 result; 250 251 result.append("Clients:\n"); 252 for (size_t i = 0; i < mClients.size(); ++i) { 253 sp<Client> client = mClients.valueAt(i).promote(); 254 if (client != 0) { 255 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 256 result.append(buffer); 257 } 258 } 259 260 result.append("Global session refs:\n"); 261 result.append(" session pid count\n"); 262 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 263 AudioSessionRef *r = mAudioSessionRefs[i]; 264 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 265 result.append(buffer); 266 } 267 write(fd, result.string(), result.size()); 268 return NO_ERROR; 269} 270 271 272status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 273{ 274 const size_t SIZE = 256; 275 char buffer[SIZE]; 276 String8 result; 277 hardware_call_state hardwareStatus = mHardwareStatus; 278 279 snprintf(buffer, SIZE, "Hardware status: %d\n" 280 "Standby Time mSec: %u\n", 281 hardwareStatus, 282 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 283 result.append(buffer); 284 write(fd, result.string(), result.size()); 285 return NO_ERROR; 286} 287 288status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 289{ 290 const size_t SIZE = 256; 291 char buffer[SIZE]; 292 String8 result; 293 snprintf(buffer, SIZE, "Permission Denial: " 294 "can't dump AudioFlinger from pid=%d, uid=%d\n", 295 IPCThreadState::self()->getCallingPid(), 296 IPCThreadState::self()->getCallingUid()); 297 result.append(buffer); 298 write(fd, result.string(), result.size()); 299 return NO_ERROR; 300} 301 302static bool tryLock(Mutex& mutex) 303{ 304 bool locked = false; 305 for (int i = 0; i < kDumpLockRetries; ++i) { 306 if (mutex.tryLock() == NO_ERROR) { 307 locked = true; 308 break; 309 } 310 usleep(kDumpLockSleepUs); 311 } 312 return locked; 313} 314 315status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 316{ 317 if (!dumpAllowed()) { 318 dumpPermissionDenial(fd, args); 319 } else { 320 // get state of hardware lock 321 bool hardwareLocked = tryLock(mHardwareLock); 322 if (!hardwareLocked) { 323 String8 result(kHardwareLockedString); 324 write(fd, result.string(), result.size()); 325 } else { 326 mHardwareLock.unlock(); 327 } 328 329 bool locked = tryLock(mLock); 330 331 // failed to lock - AudioFlinger is probably deadlocked 332 if (!locked) { 333 String8 result(kDeadlockedString); 334 write(fd, result.string(), result.size()); 335 } 336 337 dumpClients(fd, args); 338 dumpInternals(fd, args); 339 340 // dump playback threads 341 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 342 mPlaybackThreads.valueAt(i)->dump(fd, args); 343 } 344 345 // dump record threads 346 for (size_t i = 0; i < mRecordThreads.size(); i++) { 347 mRecordThreads.valueAt(i)->dump(fd, args); 348 } 349 350 // dump all hardware devs 351 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 352 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 353 dev->dump(dev, fd); 354 } 355 if (locked) mLock.unlock(); 356 } 357 return NO_ERROR; 358} 359 360sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 361{ 362 // If pid is already in the mClients wp<> map, then use that entry 363 // (for which promote() is always != 0), otherwise create a new entry and Client. 364 sp<Client> client = mClients.valueFor(pid).promote(); 365 if (client == 0) { 366 client = new Client(this, pid); 367 mClients.add(pid, client); 368 } 369 370 return client; 371} 372 373// IAudioFlinger interface 374 375 376sp<IAudioTrack> AudioFlinger::createTrack( 377 pid_t pid, 378 audio_stream_type_t streamType, 379 uint32_t sampleRate, 380 audio_format_t format, 381 uint32_t channelMask, 382 int frameCount, 383 IAudioFlinger::track_flags_t flags, 384 const sp<IMemory>& sharedBuffer, 385 audio_io_handle_t output, 386 int *sessionId, 387 status_t *status) 388{ 389 sp<PlaybackThread::Track> track; 390 sp<TrackHandle> trackHandle; 391 sp<Client> client; 392 status_t lStatus; 393 int lSessionId; 394 395 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 396 // but if someone uses binder directly they could bypass that and cause us to crash 397 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 398 ALOGE("createTrack() invalid stream type %d", streamType); 399 lStatus = BAD_VALUE; 400 goto Exit; 401 } 402 403 { 404 Mutex::Autolock _l(mLock); 405 PlaybackThread *thread = checkPlaybackThread_l(output); 406 PlaybackThread *effectThread = NULL; 407 if (thread == NULL) { 408 ALOGE("unknown output thread"); 409 lStatus = BAD_VALUE; 410 goto Exit; 411 } 412 413 client = registerPid_l(pid); 414 415 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 416 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 417 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 418 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 419 if (mPlaybackThreads.keyAt(i) != output) { 420 // prevent same audio session on different output threads 421 uint32_t sessions = t->hasAudioSession(*sessionId); 422 if (sessions & PlaybackThread::TRACK_SESSION) { 423 ALOGE("createTrack() session ID %d already in use", *sessionId); 424 lStatus = BAD_VALUE; 425 goto Exit; 426 } 427 // check if an effect with same session ID is waiting for a track to be created 428 if (sessions & PlaybackThread::EFFECT_SESSION) { 429 effectThread = t.get(); 430 } 431 } 432 } 433 lSessionId = *sessionId; 434 } else { 435 // if no audio session id is provided, create one here 436 lSessionId = nextUniqueId(); 437 if (sessionId != NULL) { 438 *sessionId = lSessionId; 439 } 440 } 441 ALOGV("createTrack() lSessionId: %d", lSessionId); 442 443 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 444 track = thread->createTrack_l(client, streamType, sampleRate, format, 445 channelMask, frameCount, sharedBuffer, lSessionId, flags, &lStatus); 446 447 // move effect chain to this output thread if an effect on same session was waiting 448 // for a track to be created 449 if (lStatus == NO_ERROR && effectThread != NULL) { 450 Mutex::Autolock _dl(thread->mLock); 451 Mutex::Autolock _sl(effectThread->mLock); 452 moveEffectChain_l(lSessionId, effectThread, thread, true); 453 } 454 455 // Look for sync events awaiting for a session to be used. 456 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 457 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 458 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 459 track->setSyncEvent(mPendingSyncEvents[i]); 460 mPendingSyncEvents.removeAt(i); 461 i--; 462 } 463 } 464 } 465 } 466 if (lStatus == NO_ERROR) { 467 trackHandle = new TrackHandle(track); 468 } else { 469 // remove local strong reference to Client before deleting the Track so that the Client 470 // destructor is called by the TrackBase destructor with mLock held 471 client.clear(); 472 track.clear(); 473 } 474 475Exit: 476 if (status != NULL) { 477 *status = lStatus; 478 } 479 return trackHandle; 480} 481 482uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 483{ 484 Mutex::Autolock _l(mLock); 485 PlaybackThread *thread = checkPlaybackThread_l(output); 486 if (thread == NULL) { 487 ALOGW("sampleRate() unknown thread %d", output); 488 return 0; 489 } 490 return thread->sampleRate(); 491} 492 493int AudioFlinger::channelCount(audio_io_handle_t output) const 494{ 495 Mutex::Autolock _l(mLock); 496 PlaybackThread *thread = checkPlaybackThread_l(output); 497 if (thread == NULL) { 498 ALOGW("channelCount() unknown thread %d", output); 499 return 0; 500 } 501 return thread->channelCount(); 502} 503 504audio_format_t AudioFlinger::format(audio_io_handle_t output) const 505{ 506 Mutex::Autolock _l(mLock); 507 PlaybackThread *thread = checkPlaybackThread_l(output); 508 if (thread == NULL) { 509 ALOGW("format() unknown thread %d", output); 510 return AUDIO_FORMAT_INVALID; 511 } 512 return thread->format(); 513} 514 515size_t AudioFlinger::frameCount(audio_io_handle_t output) const 516{ 517 Mutex::Autolock _l(mLock); 518 PlaybackThread *thread = checkPlaybackThread_l(output); 519 if (thread == NULL) { 520 ALOGW("frameCount() unknown thread %d", output); 521 return 0; 522 } 523 return thread->frameCount(); 524} 525 526uint32_t AudioFlinger::latency(audio_io_handle_t output) const 527{ 528 Mutex::Autolock _l(mLock); 529 PlaybackThread *thread = checkPlaybackThread_l(output); 530 if (thread == NULL) { 531 ALOGW("latency() unknown thread %d", output); 532 return 0; 533 } 534 return thread->latency(); 535} 536 537status_t AudioFlinger::setMasterVolume(float value) 538{ 539 status_t ret = initCheck(); 540 if (ret != NO_ERROR) { 541 return ret; 542 } 543 544 // check calling permissions 545 if (!settingsAllowed()) { 546 return PERMISSION_DENIED; 547 } 548 549 float swmv = value; 550 551 Mutex::Autolock _l(mLock); 552 553 // when hw supports master volume, don't scale in sw mixer 554 if (MVS_NONE != mMasterVolumeSupportLvl) { 555 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 556 AutoMutex lock(mHardwareLock); 557 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 558 559 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 560 if (NULL != dev->set_master_volume) { 561 dev->set_master_volume(dev, value); 562 } 563 mHardwareStatus = AUDIO_HW_IDLE; 564 } 565 566 swmv = 1.0; 567 } 568 569 mMasterVolume = value; 570 mMasterVolumeSW = swmv; 571 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 572 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 573 574 return NO_ERROR; 575} 576 577status_t AudioFlinger::setMode(audio_mode_t mode) 578{ 579 status_t ret = initCheck(); 580 if (ret != NO_ERROR) { 581 return ret; 582 } 583 584 // check calling permissions 585 if (!settingsAllowed()) { 586 return PERMISSION_DENIED; 587 } 588 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 589 ALOGW("Illegal value: setMode(%d)", mode); 590 return BAD_VALUE; 591 } 592 593 { // scope for the lock 594 AutoMutex lock(mHardwareLock); 595 mHardwareStatus = AUDIO_HW_SET_MODE; 596 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 597 mHardwareStatus = AUDIO_HW_IDLE; 598 } 599 600 if (NO_ERROR == ret) { 601 Mutex::Autolock _l(mLock); 602 mMode = mode; 603 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 604 mPlaybackThreads.valueAt(i)->setMode(mode); 605 } 606 607 return ret; 608} 609 610status_t AudioFlinger::setMicMute(bool state) 611{ 612 status_t ret = initCheck(); 613 if (ret != NO_ERROR) { 614 return ret; 615 } 616 617 // check calling permissions 618 if (!settingsAllowed()) { 619 return PERMISSION_DENIED; 620 } 621 622 AutoMutex lock(mHardwareLock); 623 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 624 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 625 mHardwareStatus = AUDIO_HW_IDLE; 626 return ret; 627} 628 629bool AudioFlinger::getMicMute() const 630{ 631 status_t ret = initCheck(); 632 if (ret != NO_ERROR) { 633 return false; 634 } 635 636 bool state = AUDIO_MODE_INVALID; 637 AutoMutex lock(mHardwareLock); 638 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 639 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 640 mHardwareStatus = AUDIO_HW_IDLE; 641 return state; 642} 643 644status_t AudioFlinger::setMasterMute(bool muted) 645{ 646 // check calling permissions 647 if (!settingsAllowed()) { 648 return PERMISSION_DENIED; 649 } 650 651 Mutex::Autolock _l(mLock); 652 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 653 mMasterMute = muted; 654 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 655 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 656 657 return NO_ERROR; 658} 659 660float AudioFlinger::masterVolume() const 661{ 662 Mutex::Autolock _l(mLock); 663 return masterVolume_l(); 664} 665 666float AudioFlinger::masterVolumeSW() const 667{ 668 Mutex::Autolock _l(mLock); 669 return masterVolumeSW_l(); 670} 671 672bool AudioFlinger::masterMute() const 673{ 674 Mutex::Autolock _l(mLock); 675 return masterMute_l(); 676} 677 678float AudioFlinger::masterVolume_l() const 679{ 680 if (MVS_FULL == mMasterVolumeSupportLvl) { 681 float ret_val; 682 AutoMutex lock(mHardwareLock); 683 684 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 685 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 686 (NULL != mPrimaryHardwareDev->get_master_volume), 687 "can't get master volume"); 688 689 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 690 mHardwareStatus = AUDIO_HW_IDLE; 691 return ret_val; 692 } 693 694 return mMasterVolume; 695} 696 697status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 698 audio_io_handle_t output) 699{ 700 // check calling permissions 701 if (!settingsAllowed()) { 702 return PERMISSION_DENIED; 703 } 704 705 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 706 ALOGE("setStreamVolume() invalid stream %d", stream); 707 return BAD_VALUE; 708 } 709 710 AutoMutex lock(mLock); 711 PlaybackThread *thread = NULL; 712 if (output) { 713 thread = checkPlaybackThread_l(output); 714 if (thread == NULL) { 715 return BAD_VALUE; 716 } 717 } 718 719 mStreamTypes[stream].volume = value; 720 721 if (thread == NULL) { 722 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 723 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 724 } 725 } else { 726 thread->setStreamVolume(stream, value); 727 } 728 729 return NO_ERROR; 730} 731 732status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 733{ 734 // check calling permissions 735 if (!settingsAllowed()) { 736 return PERMISSION_DENIED; 737 } 738 739 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 740 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 741 ALOGE("setStreamMute() invalid stream %d", stream); 742 return BAD_VALUE; 743 } 744 745 AutoMutex lock(mLock); 746 mStreamTypes[stream].mute = muted; 747 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 748 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 749 750 return NO_ERROR; 751} 752 753float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 754{ 755 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 756 return 0.0f; 757 } 758 759 AutoMutex lock(mLock); 760 float volume; 761 if (output) { 762 PlaybackThread *thread = checkPlaybackThread_l(output); 763 if (thread == NULL) { 764 return 0.0f; 765 } 766 volume = thread->streamVolume(stream); 767 } else { 768 volume = streamVolume_l(stream); 769 } 770 771 return volume; 772} 773 774bool AudioFlinger::streamMute(audio_stream_type_t stream) const 775{ 776 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 777 return true; 778 } 779 780 AutoMutex lock(mLock); 781 return streamMute_l(stream); 782} 783 784status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 785{ 786 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 787 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 788 // check calling permissions 789 if (!settingsAllowed()) { 790 return PERMISSION_DENIED; 791 } 792 793 // ioHandle == 0 means the parameters are global to the audio hardware interface 794 if (ioHandle == 0) { 795 Mutex::Autolock _l(mLock); 796 status_t final_result = NO_ERROR; 797 { 798 AutoMutex lock(mHardwareLock); 799 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 800 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 801 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 802 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 803 final_result = result ?: final_result; 804 } 805 mHardwareStatus = AUDIO_HW_IDLE; 806 } 807 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 808 AudioParameter param = AudioParameter(keyValuePairs); 809 String8 value; 810 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 811 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 812 if (mBtNrecIsOff != btNrecIsOff) { 813 for (size_t i = 0; i < mRecordThreads.size(); i++) { 814 sp<RecordThread> thread = mRecordThreads.valueAt(i); 815 RecordThread::RecordTrack *track = thread->track(); 816 if (track != NULL) { 817 audio_devices_t device = (audio_devices_t)( 818 thread->device() & AUDIO_DEVICE_IN_ALL); 819 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 820 thread->setEffectSuspended(FX_IID_AEC, 821 suspend, 822 track->sessionId()); 823 thread->setEffectSuspended(FX_IID_NS, 824 suspend, 825 track->sessionId()); 826 } 827 } 828 mBtNrecIsOff = btNrecIsOff; 829 } 830 } 831 return final_result; 832 } 833 834 // hold a strong ref on thread in case closeOutput() or closeInput() is called 835 // and the thread is exited once the lock is released 836 sp<ThreadBase> thread; 837 { 838 Mutex::Autolock _l(mLock); 839 thread = checkPlaybackThread_l(ioHandle); 840 if (thread == NULL) { 841 thread = checkRecordThread_l(ioHandle); 842 } else if (thread == primaryPlaybackThread_l()) { 843 // indicate output device change to all input threads for pre processing 844 AudioParameter param = AudioParameter(keyValuePairs); 845 int value; 846 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 847 (value != 0)) { 848 for (size_t i = 0; i < mRecordThreads.size(); i++) { 849 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 850 } 851 } 852 } 853 } 854 if (thread != 0) { 855 return thread->setParameters(keyValuePairs); 856 } 857 return BAD_VALUE; 858} 859 860String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 861{ 862// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 863// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 864 865 Mutex::Autolock _l(mLock); 866 867 if (ioHandle == 0) { 868 String8 out_s8; 869 870 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 871 char *s; 872 { 873 AutoMutex lock(mHardwareLock); 874 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 875 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 876 s = dev->get_parameters(dev, keys.string()); 877 mHardwareStatus = AUDIO_HW_IDLE; 878 } 879 out_s8 += String8(s ? s : ""); 880 free(s); 881 } 882 return out_s8; 883 } 884 885 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 886 if (playbackThread != NULL) { 887 return playbackThread->getParameters(keys); 888 } 889 RecordThread *recordThread = checkRecordThread_l(ioHandle); 890 if (recordThread != NULL) { 891 return recordThread->getParameters(keys); 892 } 893 return String8(""); 894} 895 896size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 897{ 898 status_t ret = initCheck(); 899 if (ret != NO_ERROR) { 900 return 0; 901 } 902 903 AutoMutex lock(mHardwareLock); 904 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 905 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 906 mHardwareStatus = AUDIO_HW_IDLE; 907 return size; 908} 909 910unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 911{ 912 if (ioHandle == 0) { 913 return 0; 914 } 915 916 Mutex::Autolock _l(mLock); 917 918 RecordThread *recordThread = checkRecordThread_l(ioHandle); 919 if (recordThread != NULL) { 920 return recordThread->getInputFramesLost(); 921 } 922 return 0; 923} 924 925status_t AudioFlinger::setVoiceVolume(float value) 926{ 927 status_t ret = initCheck(); 928 if (ret != NO_ERROR) { 929 return ret; 930 } 931 932 // check calling permissions 933 if (!settingsAllowed()) { 934 return PERMISSION_DENIED; 935 } 936 937 AutoMutex lock(mHardwareLock); 938 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 939 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 940 mHardwareStatus = AUDIO_HW_IDLE; 941 942 return ret; 943} 944 945status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 946 audio_io_handle_t output) const 947{ 948 status_t status; 949 950 Mutex::Autolock _l(mLock); 951 952 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 953 if (playbackThread != NULL) { 954 return playbackThread->getRenderPosition(halFrames, dspFrames); 955 } 956 957 return BAD_VALUE; 958} 959 960void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 961{ 962 963 Mutex::Autolock _l(mLock); 964 965 pid_t pid = IPCThreadState::self()->getCallingPid(); 966 if (mNotificationClients.indexOfKey(pid) < 0) { 967 sp<NotificationClient> notificationClient = new NotificationClient(this, 968 client, 969 pid); 970 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 971 972 mNotificationClients.add(pid, notificationClient); 973 974 sp<IBinder> binder = client->asBinder(); 975 binder->linkToDeath(notificationClient); 976 977 // the config change is always sent from playback or record threads to avoid deadlock 978 // with AudioSystem::gLock 979 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 980 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 981 } 982 983 for (size_t i = 0; i < mRecordThreads.size(); i++) { 984 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 985 } 986 } 987} 988 989void AudioFlinger::removeNotificationClient(pid_t pid) 990{ 991 Mutex::Autolock _l(mLock); 992 993 mNotificationClients.removeItem(pid); 994 995 ALOGV("%d died, releasing its sessions", pid); 996 size_t num = mAudioSessionRefs.size(); 997 bool removed = false; 998 for (size_t i = 0; i< num; ) { 999 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1000 ALOGV(" pid %d @ %d", ref->mPid, i); 1001 if (ref->mPid == pid) { 1002 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1003 mAudioSessionRefs.removeAt(i); 1004 delete ref; 1005 removed = true; 1006 num--; 1007 } else { 1008 i++; 1009 } 1010 } 1011 if (removed) { 1012 purgeStaleEffects_l(); 1013 } 1014} 1015 1016// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1017void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1018{ 1019 size_t size = mNotificationClients.size(); 1020 for (size_t i = 0; i < size; i++) { 1021 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1022 param2); 1023 } 1024} 1025 1026// removeClient_l() must be called with AudioFlinger::mLock held 1027void AudioFlinger::removeClient_l(pid_t pid) 1028{ 1029 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1030 mClients.removeItem(pid); 1031} 1032 1033 1034// ---------------------------------------------------------------------------- 1035 1036AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1037 uint32_t device, type_t type) 1038 : Thread(false), 1039 mType(type), 1040 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1041 // mChannelMask 1042 mChannelCount(0), 1043 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1044 mParamStatus(NO_ERROR), 1045 mStandby(false), mId(id), 1046 mDevice(device), 1047 mDeathRecipient(new PMDeathRecipient(this)) 1048{ 1049} 1050 1051AudioFlinger::ThreadBase::~ThreadBase() 1052{ 1053 mParamCond.broadcast(); 1054 // do not lock the mutex in destructor 1055 releaseWakeLock_l(); 1056 if (mPowerManager != 0) { 1057 sp<IBinder> binder = mPowerManager->asBinder(); 1058 binder->unlinkToDeath(mDeathRecipient); 1059 } 1060} 1061 1062void AudioFlinger::ThreadBase::exit() 1063{ 1064 ALOGV("ThreadBase::exit"); 1065 { 1066 // This lock prevents the following race in thread (uniprocessor for illustration): 1067 // if (!exitPending()) { 1068 // // context switch from here to exit() 1069 // // exit() calls requestExit(), what exitPending() observes 1070 // // exit() calls signal(), which is dropped since no waiters 1071 // // context switch back from exit() to here 1072 // mWaitWorkCV.wait(...); 1073 // // now thread is hung 1074 // } 1075 AutoMutex lock(mLock); 1076 requestExit(); 1077 mWaitWorkCV.signal(); 1078 } 1079 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1080 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1081 requestExitAndWait(); 1082} 1083 1084status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1085{ 1086 status_t status; 1087 1088 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1089 Mutex::Autolock _l(mLock); 1090 1091 mNewParameters.add(keyValuePairs); 1092 mWaitWorkCV.signal(); 1093 // wait condition with timeout in case the thread loop has exited 1094 // before the request could be processed 1095 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1096 status = mParamStatus; 1097 mWaitWorkCV.signal(); 1098 } else { 1099 status = TIMED_OUT; 1100 } 1101 return status; 1102} 1103 1104void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1105{ 1106 Mutex::Autolock _l(mLock); 1107 sendConfigEvent_l(event, param); 1108} 1109 1110// sendConfigEvent_l() must be called with ThreadBase::mLock held 1111void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1112{ 1113 ConfigEvent configEvent; 1114 configEvent.mEvent = event; 1115 configEvent.mParam = param; 1116 mConfigEvents.add(configEvent); 1117 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1118 mWaitWorkCV.signal(); 1119} 1120 1121void AudioFlinger::ThreadBase::processConfigEvents() 1122{ 1123 mLock.lock(); 1124 while (!mConfigEvents.isEmpty()) { 1125 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1126 ConfigEvent configEvent = mConfigEvents[0]; 1127 mConfigEvents.removeAt(0); 1128 // release mLock before locking AudioFlinger mLock: lock order is always 1129 // AudioFlinger then ThreadBase to avoid cross deadlock 1130 mLock.unlock(); 1131 mAudioFlinger->mLock.lock(); 1132 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1133 mAudioFlinger->mLock.unlock(); 1134 mLock.lock(); 1135 } 1136 mLock.unlock(); 1137} 1138 1139status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1140{ 1141 const size_t SIZE = 256; 1142 char buffer[SIZE]; 1143 String8 result; 1144 1145 bool locked = tryLock(mLock); 1146 if (!locked) { 1147 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1148 write(fd, buffer, strlen(buffer)); 1149 } 1150 1151 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1152 result.append(buffer); 1153 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1154 result.append(buffer); 1155 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1156 result.append(buffer); 1157 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1158 result.append(buffer); 1159 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1160 result.append(buffer); 1161 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1162 result.append(buffer); 1163 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1164 result.append(buffer); 1165 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1166 result.append(buffer); 1167 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1168 result.append(buffer); 1169 1170 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1171 result.append(buffer); 1172 result.append(" Index Command"); 1173 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1174 snprintf(buffer, SIZE, "\n %02d ", i); 1175 result.append(buffer); 1176 result.append(mNewParameters[i]); 1177 } 1178 1179 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1180 result.append(buffer); 1181 snprintf(buffer, SIZE, " Index event param\n"); 1182 result.append(buffer); 1183 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1184 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1185 result.append(buffer); 1186 } 1187 result.append("\n"); 1188 1189 write(fd, result.string(), result.size()); 1190 1191 if (locked) { 1192 mLock.unlock(); 1193 } 1194 return NO_ERROR; 1195} 1196 1197status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1198{ 1199 const size_t SIZE = 256; 1200 char buffer[SIZE]; 1201 String8 result; 1202 1203 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1204 write(fd, buffer, strlen(buffer)); 1205 1206 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1207 sp<EffectChain> chain = mEffectChains[i]; 1208 if (chain != 0) { 1209 chain->dump(fd, args); 1210 } 1211 } 1212 return NO_ERROR; 1213} 1214 1215void AudioFlinger::ThreadBase::acquireWakeLock() 1216{ 1217 Mutex::Autolock _l(mLock); 1218 acquireWakeLock_l(); 1219} 1220 1221void AudioFlinger::ThreadBase::acquireWakeLock_l() 1222{ 1223 if (mPowerManager == 0) { 1224 // use checkService() to avoid blocking if power service is not up yet 1225 sp<IBinder> binder = 1226 defaultServiceManager()->checkService(String16("power")); 1227 if (binder == 0) { 1228 ALOGW("Thread %s cannot connect to the power manager service", mName); 1229 } else { 1230 mPowerManager = interface_cast<IPowerManager>(binder); 1231 binder->linkToDeath(mDeathRecipient); 1232 } 1233 } 1234 if (mPowerManager != 0) { 1235 sp<IBinder> binder = new BBinder(); 1236 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1237 binder, 1238 String16(mName)); 1239 if (status == NO_ERROR) { 1240 mWakeLockToken = binder; 1241 } 1242 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1243 } 1244} 1245 1246void AudioFlinger::ThreadBase::releaseWakeLock() 1247{ 1248 Mutex::Autolock _l(mLock); 1249 releaseWakeLock_l(); 1250} 1251 1252void AudioFlinger::ThreadBase::releaseWakeLock_l() 1253{ 1254 if (mWakeLockToken != 0) { 1255 ALOGV("releaseWakeLock_l() %s", mName); 1256 if (mPowerManager != 0) { 1257 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1258 } 1259 mWakeLockToken.clear(); 1260 } 1261} 1262 1263void AudioFlinger::ThreadBase::clearPowerManager() 1264{ 1265 Mutex::Autolock _l(mLock); 1266 releaseWakeLock_l(); 1267 mPowerManager.clear(); 1268} 1269 1270void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1271{ 1272 sp<ThreadBase> thread = mThread.promote(); 1273 if (thread != 0) { 1274 thread->clearPowerManager(); 1275 } 1276 ALOGW("power manager service died !!!"); 1277} 1278 1279void AudioFlinger::ThreadBase::setEffectSuspended( 1280 const effect_uuid_t *type, bool suspend, int sessionId) 1281{ 1282 Mutex::Autolock _l(mLock); 1283 setEffectSuspended_l(type, suspend, sessionId); 1284} 1285 1286void AudioFlinger::ThreadBase::setEffectSuspended_l( 1287 const effect_uuid_t *type, bool suspend, int sessionId) 1288{ 1289 sp<EffectChain> chain = getEffectChain_l(sessionId); 1290 if (chain != 0) { 1291 if (type != NULL) { 1292 chain->setEffectSuspended_l(type, suspend); 1293 } else { 1294 chain->setEffectSuspendedAll_l(suspend); 1295 } 1296 } 1297 1298 updateSuspendedSessions_l(type, suspend, sessionId); 1299} 1300 1301void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1302{ 1303 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1304 if (index < 0) { 1305 return; 1306 } 1307 1308 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1309 mSuspendedSessions.editValueAt(index); 1310 1311 for (size_t i = 0; i < sessionEffects.size(); i++) { 1312 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1313 for (int j = 0; j < desc->mRefCount; j++) { 1314 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1315 chain->setEffectSuspendedAll_l(true); 1316 } else { 1317 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1318 desc->mType.timeLow); 1319 chain->setEffectSuspended_l(&desc->mType, true); 1320 } 1321 } 1322 } 1323} 1324 1325void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1326 bool suspend, 1327 int sessionId) 1328{ 1329 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1330 1331 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1332 1333 if (suspend) { 1334 if (index >= 0) { 1335 sessionEffects = mSuspendedSessions.editValueAt(index); 1336 } else { 1337 mSuspendedSessions.add(sessionId, sessionEffects); 1338 } 1339 } else { 1340 if (index < 0) { 1341 return; 1342 } 1343 sessionEffects = mSuspendedSessions.editValueAt(index); 1344 } 1345 1346 1347 int key = EffectChain::kKeyForSuspendAll; 1348 if (type != NULL) { 1349 key = type->timeLow; 1350 } 1351 index = sessionEffects.indexOfKey(key); 1352 1353 sp<SuspendedSessionDesc> desc; 1354 if (suspend) { 1355 if (index >= 0) { 1356 desc = sessionEffects.valueAt(index); 1357 } else { 1358 desc = new SuspendedSessionDesc(); 1359 if (type != NULL) { 1360 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1361 } 1362 sessionEffects.add(key, desc); 1363 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1364 } 1365 desc->mRefCount++; 1366 } else { 1367 if (index < 0) { 1368 return; 1369 } 1370 desc = sessionEffects.valueAt(index); 1371 if (--desc->mRefCount == 0) { 1372 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1373 sessionEffects.removeItemsAt(index); 1374 if (sessionEffects.isEmpty()) { 1375 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1376 sessionId); 1377 mSuspendedSessions.removeItem(sessionId); 1378 } 1379 } 1380 } 1381 if (!sessionEffects.isEmpty()) { 1382 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1383 } 1384} 1385 1386void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1387 bool enabled, 1388 int sessionId) 1389{ 1390 Mutex::Autolock _l(mLock); 1391 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1392} 1393 1394void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1395 bool enabled, 1396 int sessionId) 1397{ 1398 if (mType != RECORD) { 1399 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1400 // another session. This gives the priority to well behaved effect control panels 1401 // and applications not using global effects. 1402 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1403 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1404 } 1405 } 1406 1407 sp<EffectChain> chain = getEffectChain_l(sessionId); 1408 if (chain != 0) { 1409 chain->checkSuspendOnEffectEnabled(effect, enabled); 1410 } 1411} 1412 1413// ---------------------------------------------------------------------------- 1414 1415AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1416 AudioStreamOut* output, 1417 audio_io_handle_t id, 1418 uint32_t device, 1419 type_t type) 1420 : ThreadBase(audioFlinger, id, device, type), 1421 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1422 // Assumes constructor is called by AudioFlinger with it's mLock held, 1423 // but it would be safer to explicitly pass initial masterMute as parameter 1424 mMasterMute(audioFlinger->masterMute_l()), 1425 // mStreamTypes[] initialized in constructor body 1426 mOutput(output), 1427 // Assumes constructor is called by AudioFlinger with it's mLock held, 1428 // but it would be safer to explicitly pass initial masterVolume as parameter 1429 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1430 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1431 mMixerStatus(MIXER_IDLE), 1432 mPrevMixerStatus(MIXER_IDLE), 1433 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1434{ 1435 snprintf(mName, kNameLength, "AudioOut_%X", id); 1436 1437 readOutputParameters(); 1438 1439 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1440 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1441 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1442 stream = (audio_stream_type_t) (stream + 1)) { 1443 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1444 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1445 } 1446 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1447 // because mAudioFlinger doesn't have one to copy from 1448} 1449 1450AudioFlinger::PlaybackThread::~PlaybackThread() 1451{ 1452 delete [] mMixBuffer; 1453} 1454 1455status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1456{ 1457 dumpInternals(fd, args); 1458 dumpTracks(fd, args); 1459 dumpEffectChains(fd, args); 1460 return NO_ERROR; 1461} 1462 1463status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1464{ 1465 const size_t SIZE = 256; 1466 char buffer[SIZE]; 1467 String8 result; 1468 1469 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1470 result.append(buffer); 1471 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1472 for (size_t i = 0; i < mTracks.size(); ++i) { 1473 sp<Track> track = mTracks[i]; 1474 if (track != 0) { 1475 track->dump(buffer, SIZE); 1476 result.append(buffer); 1477 } 1478 } 1479 1480 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1481 result.append(buffer); 1482 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1483 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1484 sp<Track> track = mActiveTracks[i].promote(); 1485 if (track != 0) { 1486 track->dump(buffer, SIZE); 1487 result.append(buffer); 1488 } 1489 } 1490 write(fd, result.string(), result.size()); 1491 return NO_ERROR; 1492} 1493 1494status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1495{ 1496 const size_t SIZE = 256; 1497 char buffer[SIZE]; 1498 String8 result; 1499 1500 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1501 result.append(buffer); 1502 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1503 result.append(buffer); 1504 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1505 result.append(buffer); 1506 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1507 result.append(buffer); 1508 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1509 result.append(buffer); 1510 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1511 result.append(buffer); 1512 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1513 result.append(buffer); 1514 write(fd, result.string(), result.size()); 1515 1516 dumpBase(fd, args); 1517 1518 return NO_ERROR; 1519} 1520 1521// Thread virtuals 1522status_t AudioFlinger::PlaybackThread::readyToRun() 1523{ 1524 status_t status = initCheck(); 1525 if (status == NO_ERROR) { 1526 ALOGI("AudioFlinger's thread %p ready to run", this); 1527 } else { 1528 ALOGE("No working audio driver found."); 1529 } 1530 return status; 1531} 1532 1533void AudioFlinger::PlaybackThread::onFirstRef() 1534{ 1535 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1536} 1537 1538// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1539sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1540 const sp<AudioFlinger::Client>& client, 1541 audio_stream_type_t streamType, 1542 uint32_t sampleRate, 1543 audio_format_t format, 1544 uint32_t channelMask, 1545 int frameCount, 1546 const sp<IMemory>& sharedBuffer, 1547 int sessionId, 1548 IAudioFlinger::track_flags_t flags, 1549 status_t *status) 1550{ 1551 sp<Track> track; 1552 status_t lStatus; 1553 1554 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1555 1556 // client expresses a preference for FAST, but we get the final say 1557 if ((flags & IAudioFlinger::TRACK_FAST) && 1558 !( 1559 // not timed 1560 (!isTimed) && 1561 // either of these use cases: 1562 ( 1563 // use case 1: shared buffer with any frame count 1564 ( 1565 (sharedBuffer != 0) 1566 ) || 1567 // use case 2: callback handler and small power-of-2 frame count 1568 ( 1569 // unfortunately we can't verify that there's a callback until start() 1570 // FIXME supported frame counts should not be hard-coded 1571 ( 1572 (frameCount == 128) || 1573 (frameCount == 256) || 1574 (frameCount == 512) 1575 ) 1576 ) 1577 ) && 1578 // PCM data 1579 audio_is_linear_pcm(format) && 1580 // mono or stereo 1581 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1582 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1583 // hardware sample rate 1584 (sampleRate == mSampleRate) 1585 // FIXME test that MixerThread for this fast track has a capable output HAL 1586 // FIXME add a permission test also? 1587 ) ) { 1588 ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied"); 1589 flags &= ~IAudioFlinger::TRACK_FAST; 1590 } 1591 1592 if (mType == DIRECT) { 1593 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1594 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1595 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1596 "for output %p with format %d", 1597 sampleRate, format, channelMask, mOutput, mFormat); 1598 lStatus = BAD_VALUE; 1599 goto Exit; 1600 } 1601 } 1602 } else { 1603 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1604 if (sampleRate > mSampleRate*2) { 1605 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1606 lStatus = BAD_VALUE; 1607 goto Exit; 1608 } 1609 } 1610 1611 lStatus = initCheck(); 1612 if (lStatus != NO_ERROR) { 1613 ALOGE("Audio driver not initialized."); 1614 goto Exit; 1615 } 1616 1617 { // scope for mLock 1618 Mutex::Autolock _l(mLock); 1619 1620 // all tracks in same audio session must share the same routing strategy otherwise 1621 // conflicts will happen when tracks are moved from one output to another by audio policy 1622 // manager 1623 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1624 for (size_t i = 0; i < mTracks.size(); ++i) { 1625 sp<Track> t = mTracks[i]; 1626 if (t != 0 && !t->isOutputTrack()) { 1627 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1628 if (sessionId == t->sessionId() && strategy != actual) { 1629 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1630 strategy, actual); 1631 lStatus = BAD_VALUE; 1632 goto Exit; 1633 } 1634 } 1635 } 1636 1637 if (!isTimed) { 1638 track = new Track(this, client, streamType, sampleRate, format, 1639 channelMask, frameCount, sharedBuffer, sessionId, flags); 1640 } else { 1641 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1642 channelMask, frameCount, sharedBuffer, sessionId); 1643 } 1644 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1645 lStatus = NO_MEMORY; 1646 goto Exit; 1647 } 1648 mTracks.add(track); 1649 1650 sp<EffectChain> chain = getEffectChain_l(sessionId); 1651 if (chain != 0) { 1652 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1653 track->setMainBuffer(chain->inBuffer()); 1654 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1655 chain->incTrackCnt(); 1656 } 1657 } 1658 lStatus = NO_ERROR; 1659 1660Exit: 1661 if (status) { 1662 *status = lStatus; 1663 } 1664 return track; 1665} 1666 1667uint32_t AudioFlinger::PlaybackThread::latency() const 1668{ 1669 Mutex::Autolock _l(mLock); 1670 if (initCheck() == NO_ERROR) { 1671 return mOutput->stream->get_latency(mOutput->stream); 1672 } else { 1673 return 0; 1674 } 1675} 1676 1677void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1678{ 1679 Mutex::Autolock _l(mLock); 1680 mMasterVolume = value; 1681} 1682 1683void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1684{ 1685 Mutex::Autolock _l(mLock); 1686 setMasterMute_l(muted); 1687} 1688 1689void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1690{ 1691 Mutex::Autolock _l(mLock); 1692 mStreamTypes[stream].volume = value; 1693} 1694 1695void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1696{ 1697 Mutex::Autolock _l(mLock); 1698 mStreamTypes[stream].mute = muted; 1699} 1700 1701float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1702{ 1703 Mutex::Autolock _l(mLock); 1704 return mStreamTypes[stream].volume; 1705} 1706 1707// addTrack_l() must be called with ThreadBase::mLock held 1708status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1709{ 1710 status_t status = ALREADY_EXISTS; 1711 1712 // set retry count for buffer fill 1713 track->mRetryCount = kMaxTrackStartupRetries; 1714 if (mActiveTracks.indexOf(track) < 0) { 1715 // the track is newly added, make sure it fills up all its 1716 // buffers before playing. This is to ensure the client will 1717 // effectively get the latency it requested. 1718 track->mFillingUpStatus = Track::FS_FILLING; 1719 track->mResetDone = false; 1720 mActiveTracks.add(track); 1721 if (track->mainBuffer() != mMixBuffer) { 1722 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1723 if (chain != 0) { 1724 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1725 chain->incActiveTrackCnt(); 1726 } 1727 } 1728 1729 status = NO_ERROR; 1730 } 1731 1732 ALOGV("mWaitWorkCV.broadcast"); 1733 mWaitWorkCV.broadcast(); 1734 1735 return status; 1736} 1737 1738// destroyTrack_l() must be called with ThreadBase::mLock held 1739void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1740{ 1741 track->mState = TrackBase::TERMINATED; 1742 if (mActiveTracks.indexOf(track) < 0) { 1743 removeTrack_l(track); 1744 } 1745} 1746 1747void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1748{ 1749 mTracks.remove(track); 1750 deleteTrackName_l(track->name()); 1751 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1752 if (chain != 0) { 1753 chain->decTrackCnt(); 1754 } 1755} 1756 1757String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1758{ 1759 String8 out_s8 = String8(""); 1760 char *s; 1761 1762 Mutex::Autolock _l(mLock); 1763 if (initCheck() != NO_ERROR) { 1764 return out_s8; 1765 } 1766 1767 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1768 out_s8 = String8(s); 1769 free(s); 1770 return out_s8; 1771} 1772 1773// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1774void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1775 AudioSystem::OutputDescriptor desc; 1776 void *param2 = NULL; 1777 1778 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1779 1780 switch (event) { 1781 case AudioSystem::OUTPUT_OPENED: 1782 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1783 desc.channels = mChannelMask; 1784 desc.samplingRate = mSampleRate; 1785 desc.format = mFormat; 1786 desc.frameCount = mFrameCount; 1787 desc.latency = latency(); 1788 param2 = &desc; 1789 break; 1790 1791 case AudioSystem::STREAM_CONFIG_CHANGED: 1792 param2 = ¶m; 1793 case AudioSystem::OUTPUT_CLOSED: 1794 default: 1795 break; 1796 } 1797 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1798} 1799 1800void AudioFlinger::PlaybackThread::readOutputParameters() 1801{ 1802 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1803 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1804 mChannelCount = (uint16_t)popcount(mChannelMask); 1805 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1806 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1807 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1808 1809 // FIXME - Current mixer implementation only supports stereo output: Always 1810 // Allocate a stereo buffer even if HW output is mono. 1811 delete[] mMixBuffer; 1812 mMixBuffer = new int16_t[mFrameCount * 2]; 1813 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1814 1815 // force reconfiguration of effect chains and engines to take new buffer size and audio 1816 // parameters into account 1817 // Note that mLock is not held when readOutputParameters() is called from the constructor 1818 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1819 // matter. 1820 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1821 Vector< sp<EffectChain> > effectChains = mEffectChains; 1822 for (size_t i = 0; i < effectChains.size(); i ++) { 1823 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1824 } 1825} 1826 1827status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1828{ 1829 if (halFrames == NULL || dspFrames == NULL) { 1830 return BAD_VALUE; 1831 } 1832 Mutex::Autolock _l(mLock); 1833 if (initCheck() != NO_ERROR) { 1834 return INVALID_OPERATION; 1835 } 1836 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1837 1838 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1839} 1840 1841uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1842{ 1843 Mutex::Autolock _l(mLock); 1844 uint32_t result = 0; 1845 if (getEffectChain_l(sessionId) != 0) { 1846 result = EFFECT_SESSION; 1847 } 1848 1849 for (size_t i = 0; i < mTracks.size(); ++i) { 1850 sp<Track> track = mTracks[i]; 1851 if (sessionId == track->sessionId() && 1852 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1853 result |= TRACK_SESSION; 1854 break; 1855 } 1856 } 1857 1858 return result; 1859} 1860 1861uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1862{ 1863 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1864 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1865 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1866 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1867 } 1868 for (size_t i = 0; i < mTracks.size(); i++) { 1869 sp<Track> track = mTracks[i]; 1870 if (sessionId == track->sessionId() && 1871 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1872 return AudioSystem::getStrategyForStream(track->streamType()); 1873 } 1874 } 1875 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1876} 1877 1878 1879AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1880{ 1881 Mutex::Autolock _l(mLock); 1882 return mOutput; 1883} 1884 1885AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1886{ 1887 Mutex::Autolock _l(mLock); 1888 AudioStreamOut *output = mOutput; 1889 mOutput = NULL; 1890 return output; 1891} 1892 1893// this method must always be called either with ThreadBase mLock held or inside the thread loop 1894audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1895{ 1896 if (mOutput == NULL) { 1897 return NULL; 1898 } 1899 return &mOutput->stream->common; 1900} 1901 1902uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1903{ 1904 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1905 // decoding and transfer time. So sleeping for half of the latency would likely cause 1906 // underruns 1907 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1908 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1909 } else { 1910 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1911 } 1912} 1913 1914status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1915{ 1916 if (!isValidSyncEvent(event)) { 1917 return BAD_VALUE; 1918 } 1919 1920 Mutex::Autolock _l(mLock); 1921 1922 for (size_t i = 0; i < mTracks.size(); ++i) { 1923 sp<Track> track = mTracks[i]; 1924 if (event->triggerSession() == track->sessionId()) { 1925 track->setSyncEvent(event); 1926 return NO_ERROR; 1927 } 1928 } 1929 1930 return NAME_NOT_FOUND; 1931} 1932 1933bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 1934{ 1935 switch (event->type()) { 1936 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 1937 return true; 1938 default: 1939 break; 1940 } 1941 return false; 1942} 1943 1944// ---------------------------------------------------------------------------- 1945 1946AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1947 audio_io_handle_t id, uint32_t device, type_t type) 1948 : PlaybackThread(audioFlinger, output, id, device, type) 1949{ 1950 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1951 // FIXME - Current mixer implementation only supports stereo output 1952 if (mChannelCount == 1) { 1953 ALOGE("Invalid audio hardware channel count"); 1954 } 1955} 1956 1957AudioFlinger::MixerThread::~MixerThread() 1958{ 1959 delete mAudioMixer; 1960} 1961 1962class CpuStats { 1963public: 1964 CpuStats(); 1965 void sample(const String8 &title); 1966#ifdef DEBUG_CPU_USAGE 1967private: 1968 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 1969 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 1970 1971 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 1972 1973 int mCpuNum; // thread's current CPU number 1974 int mCpukHz; // frequency of thread's current CPU in kHz 1975#endif 1976}; 1977 1978CpuStats::CpuStats() 1979#ifdef DEBUG_CPU_USAGE 1980 : mCpuNum(-1), mCpukHz(-1) 1981#endif 1982{ 1983} 1984 1985void CpuStats::sample(const String8 &title) { 1986#ifdef DEBUG_CPU_USAGE 1987 // get current thread's delta CPU time in wall clock ns 1988 double wcNs; 1989 bool valid = mCpuUsage.sampleAndEnable(wcNs); 1990 1991 // record sample for wall clock statistics 1992 if (valid) { 1993 mWcStats.sample(wcNs); 1994 } 1995 1996 // get the current CPU number 1997 int cpuNum = sched_getcpu(); 1998 1999 // get the current CPU frequency in kHz 2000 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2001 2002 // check if either CPU number or frequency changed 2003 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2004 mCpuNum = cpuNum; 2005 mCpukHz = cpukHz; 2006 // ignore sample for purposes of cycles 2007 valid = false; 2008 } 2009 2010 // if no change in CPU number or frequency, then record sample for cycle statistics 2011 if (valid && mCpukHz > 0) { 2012 double cycles = wcNs * cpukHz * 0.000001; 2013 mHzStats.sample(cycles); 2014 } 2015 2016 unsigned n = mWcStats.n(); 2017 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2018 if ((n & 127) == 1) { 2019 long long elapsed = mCpuUsage.elapsed(); 2020 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2021 double perLoop = elapsed / (double) n; 2022 double perLoop100 = perLoop * 0.01; 2023 double perLoop1k = perLoop * 0.001; 2024 double mean = mWcStats.mean(); 2025 double stddev = mWcStats.stddev(); 2026 double minimum = mWcStats.minimum(); 2027 double maximum = mWcStats.maximum(); 2028 double meanCycles = mHzStats.mean(); 2029 double stddevCycles = mHzStats.stddev(); 2030 double minCycles = mHzStats.minimum(); 2031 double maxCycles = mHzStats.maximum(); 2032 mCpuUsage.resetElapsed(); 2033 mWcStats.reset(); 2034 mHzStats.reset(); 2035 ALOGD("CPU usage for %s over past %.1f secs\n" 2036 " (%u mixer loops at %.1f mean ms per loop):\n" 2037 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2038 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2039 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2040 title.string(), 2041 elapsed * .000000001, n, perLoop * .000001, 2042 mean * .001, 2043 stddev * .001, 2044 minimum * .001, 2045 maximum * .001, 2046 mean / perLoop100, 2047 stddev / perLoop100, 2048 minimum / perLoop100, 2049 maximum / perLoop100, 2050 meanCycles / perLoop1k, 2051 stddevCycles / perLoop1k, 2052 minCycles / perLoop1k, 2053 maxCycles / perLoop1k); 2054 2055 } 2056 } 2057#endif 2058}; 2059 2060void AudioFlinger::PlaybackThread::checkSilentMode_l() 2061{ 2062 if (!mMasterMute) { 2063 char value[PROPERTY_VALUE_MAX]; 2064 if (property_get("ro.audio.silent", value, "0") > 0) { 2065 char *endptr; 2066 unsigned long ul = strtoul(value, &endptr, 0); 2067 if (*endptr == '\0' && ul != 0) { 2068 ALOGD("Silence is golden"); 2069 // The setprop command will not allow a property to be changed after 2070 // the first time it is set, so we don't have to worry about un-muting. 2071 setMasterMute_l(true); 2072 } 2073 } 2074 } 2075} 2076 2077bool AudioFlinger::PlaybackThread::threadLoop() 2078{ 2079 Vector< sp<Track> > tracksToRemove; 2080 2081 standbyTime = systemTime(); 2082 2083 // MIXER 2084 nsecs_t lastWarning = 0; 2085if (mType == MIXER) { 2086 longStandbyExit = false; 2087} 2088 2089 // DUPLICATING 2090 // FIXME could this be made local to while loop? 2091 writeFrames = 0; 2092 2093 cacheParameters_l(); 2094 sleepTime = idleSleepTime; 2095 2096if (mType == MIXER) { 2097 sleepTimeShift = 0; 2098} 2099 2100 CpuStats cpuStats; 2101 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2102 2103 acquireWakeLock(); 2104 2105 while (!exitPending()) 2106 { 2107 cpuStats.sample(myName); 2108 2109 Vector< sp<EffectChain> > effectChains; 2110 2111 processConfigEvents(); 2112 2113 { // scope for mLock 2114 2115 Mutex::Autolock _l(mLock); 2116 2117 if (checkForNewParameters_l()) { 2118 cacheParameters_l(); 2119 } 2120 2121 saveOutputTracks(); 2122 2123 // put audio hardware into standby after short delay 2124 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2125 mSuspended > 0)) { 2126 if (!mStandby) { 2127 2128 threadLoop_standby(); 2129 2130 mStandby = true; 2131 mBytesWritten = 0; 2132 } 2133 2134 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2135 // we're about to wait, flush the binder command buffer 2136 IPCThreadState::self()->flushCommands(); 2137 2138 clearOutputTracks(); 2139 2140 if (exitPending()) break; 2141 2142 releaseWakeLock_l(); 2143 // wait until we have something to do... 2144 ALOGV("%s going to sleep", myName.string()); 2145 mWaitWorkCV.wait(mLock); 2146 ALOGV("%s waking up", myName.string()); 2147 acquireWakeLock_l(); 2148 2149 mPrevMixerStatus = MIXER_IDLE; 2150 2151 checkSilentMode_l(); 2152 2153 standbyTime = systemTime() + standbyDelay; 2154 sleepTime = idleSleepTime; 2155 if (mType == MIXER) { 2156 sleepTimeShift = 0; 2157 } 2158 2159 continue; 2160 } 2161 } 2162 2163 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2164 // Shift in the new status; this could be a queue if it's 2165 // useful to filter the mixer status over several cycles. 2166 mPrevMixerStatus = mMixerStatus; 2167 mMixerStatus = newMixerStatus; 2168 2169 // prevent any changes in effect chain list and in each effect chain 2170 // during mixing and effect process as the audio buffers could be deleted 2171 // or modified if an effect is created or deleted 2172 lockEffectChains_l(effectChains); 2173 } 2174 2175 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2176 threadLoop_mix(); 2177 } else { 2178 threadLoop_sleepTime(); 2179 } 2180 2181 if (mSuspended > 0) { 2182 sleepTime = suspendSleepTimeUs(); 2183 } 2184 2185 // only process effects if we're going to write 2186 if (sleepTime == 0) { 2187 for (size_t i = 0; i < effectChains.size(); i ++) { 2188 effectChains[i]->process_l(); 2189 } 2190 } 2191 2192 // enable changes in effect chain 2193 unlockEffectChains(effectChains); 2194 2195 // sleepTime == 0 means we must write to audio hardware 2196 if (sleepTime == 0) { 2197 2198 threadLoop_write(); 2199 2200if (mType == MIXER) { 2201 // write blocked detection 2202 nsecs_t now = systemTime(); 2203 nsecs_t delta = now - mLastWriteTime; 2204 if (!mStandby && delta > maxPeriod) { 2205 mNumDelayedWrites++; 2206 if ((now - lastWarning) > kWarningThrottleNs) { 2207 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2208 ns2ms(delta), mNumDelayedWrites, this); 2209 lastWarning = now; 2210 } 2211 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2212 // a different threshold. Or completely removed for what it is worth anyway... 2213 if (mStandby) { 2214 longStandbyExit = true; 2215 } 2216 } 2217} 2218 2219 mStandby = false; 2220 } else { 2221 usleep(sleepTime); 2222 } 2223 2224 // finally let go of removed track(s), without the lock held 2225 // since we can't guarantee the destructors won't acquire that 2226 // same lock. 2227 tracksToRemove.clear(); 2228 2229 // FIXME I don't understand the need for this here; 2230 // it was in the original code but maybe the 2231 // assignment in saveOutputTracks() makes this unnecessary? 2232 clearOutputTracks(); 2233 2234 // Effect chains will be actually deleted here if they were removed from 2235 // mEffectChains list during mixing or effects processing 2236 effectChains.clear(); 2237 2238 // FIXME Note that the above .clear() is no longer necessary since effectChains 2239 // is now local to this block, but will keep it for now (at least until merge done). 2240 } 2241 2242if (mType == MIXER || mType == DIRECT) { 2243 // put output stream into standby mode 2244 if (!mStandby) { 2245 mOutput->stream->common.standby(&mOutput->stream->common); 2246 } 2247} 2248if (mType == DUPLICATING) { 2249 // for DuplicatingThread, standby mode is handled by the outputTracks 2250} 2251 2252 releaseWakeLock(); 2253 2254 ALOGV("Thread %p type %d exiting", this, mType); 2255 return false; 2256} 2257 2258// shared by MIXER and DIRECT, overridden by DUPLICATING 2259void AudioFlinger::PlaybackThread::threadLoop_write() 2260{ 2261 // FIXME rewrite to reduce number of system calls 2262 mLastWriteTime = systemTime(); 2263 mInWrite = true; 2264 mBytesWritten += mixBufferSize; 2265 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2266 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2267 mNumWrites++; 2268 mInWrite = false; 2269} 2270 2271// shared by MIXER and DIRECT, overridden by DUPLICATING 2272void AudioFlinger::PlaybackThread::threadLoop_standby() 2273{ 2274 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2275 mOutput->stream->common.standby(&mOutput->stream->common); 2276} 2277 2278void AudioFlinger::MixerThread::threadLoop_mix() 2279{ 2280 // obtain the presentation timestamp of the next output buffer 2281 int64_t pts; 2282 status_t status = INVALID_OPERATION; 2283 2284 if (NULL != mOutput->stream->get_next_write_timestamp) { 2285 status = mOutput->stream->get_next_write_timestamp( 2286 mOutput->stream, &pts); 2287 } 2288 2289 if (status != NO_ERROR) { 2290 pts = AudioBufferProvider::kInvalidPTS; 2291 } 2292 2293 // mix buffers... 2294 mAudioMixer->process(pts); 2295 // increase sleep time progressively when application underrun condition clears. 2296 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2297 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2298 // such that we would underrun the audio HAL. 2299 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2300 sleepTimeShift--; 2301 } 2302 sleepTime = 0; 2303 standbyTime = systemTime() + standbyDelay; 2304 //TODO: delay standby when effects have a tail 2305} 2306 2307void AudioFlinger::MixerThread::threadLoop_sleepTime() 2308{ 2309 // If no tracks are ready, sleep once for the duration of an output 2310 // buffer size, then write 0s to the output 2311 if (sleepTime == 0) { 2312 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2313 sleepTime = activeSleepTime >> sleepTimeShift; 2314 if (sleepTime < kMinThreadSleepTimeUs) { 2315 sleepTime = kMinThreadSleepTimeUs; 2316 } 2317 // reduce sleep time in case of consecutive application underruns to avoid 2318 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2319 // duration we would end up writing less data than needed by the audio HAL if 2320 // the condition persists. 2321 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2322 sleepTimeShift++; 2323 } 2324 } else { 2325 sleepTime = idleSleepTime; 2326 } 2327 } else if (mBytesWritten != 0 || 2328 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2329 memset (mMixBuffer, 0, mixBufferSize); 2330 sleepTime = 0; 2331 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2332 } 2333 // TODO add standby time extension fct of effect tail 2334} 2335 2336// prepareTracks_l() must be called with ThreadBase::mLock held 2337AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2338 Vector< sp<Track> > *tracksToRemove) 2339{ 2340 2341 mixer_state mixerStatus = MIXER_IDLE; 2342 // find out which tracks need to be processed 2343 size_t count = mActiveTracks.size(); 2344 size_t mixedTracks = 0; 2345 size_t tracksWithEffect = 0; 2346 2347 float masterVolume = mMasterVolume; 2348 bool masterMute = mMasterMute; 2349 2350 if (masterMute) { 2351 masterVolume = 0; 2352 } 2353 // Delegate master volume control to effect in output mix effect chain if needed 2354 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2355 if (chain != 0) { 2356 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2357 chain->setVolume_l(&v, &v); 2358 masterVolume = (float)((v + (1 << 23)) >> 24); 2359 chain.clear(); 2360 } 2361 2362 for (size_t i=0 ; i<count ; i++) { 2363 sp<Track> t = mActiveTracks[i].promote(); 2364 if (t == 0) continue; 2365 2366 // this const just means the local variable doesn't change 2367 Track* const track = t.get(); 2368 audio_track_cblk_t* cblk = track->cblk(); 2369 2370 // The first time a track is added we wait 2371 // for all its buffers to be filled before processing it 2372 int name = track->name(); 2373 // make sure that we have enough frames to mix one full buffer. 2374 // enforce this condition only once to enable draining the buffer in case the client 2375 // app does not call stop() and relies on underrun to stop: 2376 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2377 // during last round 2378 uint32_t minFrames = 1; 2379 if (!track->isStopped() && !track->isPausing() && 2380 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2381 if (t->sampleRate() == (int)mSampleRate) { 2382 minFrames = mFrameCount; 2383 } else { 2384 // +1 for rounding and +1 for additional sample needed for interpolation 2385 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2386 // add frames already consumed but not yet released by the resampler 2387 // because cblk->framesReady() will include these frames 2388 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2389 // the minimum track buffer size is normally twice the number of frames necessary 2390 // to fill one buffer and the resampler should not leave more than one buffer worth 2391 // of unreleased frames after each pass, but just in case... 2392 ALOG_ASSERT(minFrames <= cblk->frameCount); 2393 } 2394 } 2395 if ((track->framesReady() >= minFrames) && track->isReady() && 2396 !track->isPaused() && !track->isTerminated()) 2397 { 2398 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2399 2400 mixedTracks++; 2401 2402 // track->mainBuffer() != mMixBuffer means there is an effect chain 2403 // connected to the track 2404 chain.clear(); 2405 if (track->mainBuffer() != mMixBuffer) { 2406 chain = getEffectChain_l(track->sessionId()); 2407 // Delegate volume control to effect in track effect chain if needed 2408 if (chain != 0) { 2409 tracksWithEffect++; 2410 } else { 2411 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2412 name, track->sessionId()); 2413 } 2414 } 2415 2416 2417 int param = AudioMixer::VOLUME; 2418 if (track->mFillingUpStatus == Track::FS_FILLED) { 2419 // no ramp for the first volume setting 2420 track->mFillingUpStatus = Track::FS_ACTIVE; 2421 if (track->mState == TrackBase::RESUMING) { 2422 track->mState = TrackBase::ACTIVE; 2423 param = AudioMixer::RAMP_VOLUME; 2424 } 2425 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2426 } else if (cblk->server != 0) { 2427 // If the track is stopped before the first frame was mixed, 2428 // do not apply ramp 2429 param = AudioMixer::RAMP_VOLUME; 2430 } 2431 2432 // compute volume for this track 2433 uint32_t vl, vr, va; 2434 if (track->isMuted() || track->isPausing() || 2435 mStreamTypes[track->streamType()].mute) { 2436 vl = vr = va = 0; 2437 if (track->isPausing()) { 2438 track->setPaused(); 2439 } 2440 } else { 2441 2442 // read original volumes with volume control 2443 float typeVolume = mStreamTypes[track->streamType()].volume; 2444 float v = masterVolume * typeVolume; 2445 uint32_t vlr = cblk->getVolumeLR(); 2446 vl = vlr & 0xFFFF; 2447 vr = vlr >> 16; 2448 // track volumes come from shared memory, so can't be trusted and must be clamped 2449 if (vl > MAX_GAIN_INT) { 2450 ALOGV("Track left volume out of range: %04X", vl); 2451 vl = MAX_GAIN_INT; 2452 } 2453 if (vr > MAX_GAIN_INT) { 2454 ALOGV("Track right volume out of range: %04X", vr); 2455 vr = MAX_GAIN_INT; 2456 } 2457 // now apply the master volume and stream type volume 2458 vl = (uint32_t)(v * vl) << 12; 2459 vr = (uint32_t)(v * vr) << 12; 2460 // assuming master volume and stream type volume each go up to 1.0, 2461 // vl and vr are now in 8.24 format 2462 2463 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2464 // send level comes from shared memory and so may be corrupt 2465 if (sendLevel > MAX_GAIN_INT) { 2466 ALOGV("Track send level out of range: %04X", sendLevel); 2467 sendLevel = MAX_GAIN_INT; 2468 } 2469 va = (uint32_t)(v * sendLevel); 2470 } 2471 // Delegate volume control to effect in track effect chain if needed 2472 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2473 // Do not ramp volume if volume is controlled by effect 2474 param = AudioMixer::VOLUME; 2475 track->mHasVolumeController = true; 2476 } else { 2477 // force no volume ramp when volume controller was just disabled or removed 2478 // from effect chain to avoid volume spike 2479 if (track->mHasVolumeController) { 2480 param = AudioMixer::VOLUME; 2481 } 2482 track->mHasVolumeController = false; 2483 } 2484 2485 // Convert volumes from 8.24 to 4.12 format 2486 // This additional clamping is needed in case chain->setVolume_l() overshot 2487 vl = (vl + (1 << 11)) >> 12; 2488 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2489 vr = (vr + (1 << 11)) >> 12; 2490 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2491 2492 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2493 2494 // XXX: these things DON'T need to be done each time 2495 mAudioMixer->setBufferProvider(name, track); 2496 mAudioMixer->enable(name); 2497 2498 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2499 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2500 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2501 mAudioMixer->setParameter( 2502 name, 2503 AudioMixer::TRACK, 2504 AudioMixer::FORMAT, (void *)track->format()); 2505 mAudioMixer->setParameter( 2506 name, 2507 AudioMixer::TRACK, 2508 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2509 mAudioMixer->setParameter( 2510 name, 2511 AudioMixer::RESAMPLE, 2512 AudioMixer::SAMPLE_RATE, 2513 (void *)(cblk->sampleRate)); 2514 mAudioMixer->setParameter( 2515 name, 2516 AudioMixer::TRACK, 2517 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2518 mAudioMixer->setParameter( 2519 name, 2520 AudioMixer::TRACK, 2521 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2522 2523 // reset retry count 2524 track->mRetryCount = kMaxTrackRetries; 2525 2526 // If one track is ready, set the mixer ready if: 2527 // - the mixer was not ready during previous round OR 2528 // - no other track is not ready 2529 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2530 mixerStatus != MIXER_TRACKS_ENABLED) { 2531 mixerStatus = MIXER_TRACKS_READY; 2532 } 2533 } else { 2534 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2535 if (track->isStopped()) { 2536 track->reset(); 2537 } 2538 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2539 // We have consumed all the buffers of this track. 2540 // Remove it from the list of active tracks. 2541 // TODO: use actual buffer filling status instead of latency when available from 2542 // audio HAL 2543 size_t audioHALFrames = 2544 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2545 size_t framesWritten = 2546 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2547 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2548 tracksToRemove->add(track); 2549 } 2550 } else { 2551 // No buffers for this track. Give it a few chances to 2552 // fill a buffer, then remove it from active list. 2553 if (--(track->mRetryCount) <= 0) { 2554 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2555 tracksToRemove->add(track); 2556 // indicate to client process that the track was disabled because of underrun 2557 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2558 // If one track is not ready, mark the mixer also not ready if: 2559 // - the mixer was ready during previous round OR 2560 // - no other track is ready 2561 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2562 mixerStatus != MIXER_TRACKS_READY) { 2563 mixerStatus = MIXER_TRACKS_ENABLED; 2564 } 2565 } 2566 mAudioMixer->disable(name); 2567 } 2568 } 2569 2570 // remove all the tracks that need to be... 2571 count = tracksToRemove->size(); 2572 if (CC_UNLIKELY(count)) { 2573 for (size_t i=0 ; i<count ; i++) { 2574 const sp<Track>& track = tracksToRemove->itemAt(i); 2575 mActiveTracks.remove(track); 2576 if (track->mainBuffer() != mMixBuffer) { 2577 chain = getEffectChain_l(track->sessionId()); 2578 if (chain != 0) { 2579 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2580 chain->decActiveTrackCnt(); 2581 } 2582 } 2583 if (track->isTerminated()) { 2584 removeTrack_l(track); 2585 } 2586 } 2587 } 2588 2589 // mix buffer must be cleared if all tracks are connected to an 2590 // effect chain as in this case the mixer will not write to 2591 // mix buffer and track effects will accumulate into it 2592 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2593 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2594 } 2595 2596 return mixerStatus; 2597} 2598 2599/* 2600The derived values that are cached: 2601 - mixBufferSize from frame count * frame size 2602 - activeSleepTime from activeSleepTimeUs() 2603 - idleSleepTime from idleSleepTimeUs() 2604 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2605 - maxPeriod from frame count and sample rate (MIXER only) 2606 2607The parameters that affect these derived values are: 2608 - frame count 2609 - frame size 2610 - sample rate 2611 - device type: A2DP or not 2612 - device latency 2613 - format: PCM or not 2614 - active sleep time 2615 - idle sleep time 2616*/ 2617 2618void AudioFlinger::PlaybackThread::cacheParameters_l() 2619{ 2620 mixBufferSize = mFrameCount * mFrameSize; 2621 activeSleepTime = activeSleepTimeUs(); 2622 idleSleepTime = idleSleepTimeUs(); 2623} 2624 2625void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2626{ 2627 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2628 this, streamType, mTracks.size()); 2629 Mutex::Autolock _l(mLock); 2630 2631 size_t size = mTracks.size(); 2632 for (size_t i = 0; i < size; i++) { 2633 sp<Track> t = mTracks[i]; 2634 if (t->streamType() == streamType) { 2635 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2636 t->mCblk->cv.signal(); 2637 } 2638 } 2639} 2640 2641// getTrackName_l() must be called with ThreadBase::mLock held 2642int AudioFlinger::MixerThread::getTrackName_l() 2643{ 2644 return mAudioMixer->getTrackName(); 2645} 2646 2647// deleteTrackName_l() must be called with ThreadBase::mLock held 2648void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2649{ 2650 ALOGV("remove track (%d) and delete from mixer", name); 2651 mAudioMixer->deleteTrackName(name); 2652} 2653 2654// checkForNewParameters_l() must be called with ThreadBase::mLock held 2655bool AudioFlinger::MixerThread::checkForNewParameters_l() 2656{ 2657 bool reconfig = false; 2658 2659 while (!mNewParameters.isEmpty()) { 2660 status_t status = NO_ERROR; 2661 String8 keyValuePair = mNewParameters[0]; 2662 AudioParameter param = AudioParameter(keyValuePair); 2663 int value; 2664 2665 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2666 reconfig = true; 2667 } 2668 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2669 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2670 status = BAD_VALUE; 2671 } else { 2672 reconfig = true; 2673 } 2674 } 2675 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2676 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2677 status = BAD_VALUE; 2678 } else { 2679 reconfig = true; 2680 } 2681 } 2682 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2683 // do not accept frame count changes if tracks are open as the track buffer 2684 // size depends on frame count and correct behavior would not be guaranteed 2685 // if frame count is changed after track creation 2686 if (!mTracks.isEmpty()) { 2687 status = INVALID_OPERATION; 2688 } else { 2689 reconfig = true; 2690 } 2691 } 2692 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2693#ifdef ADD_BATTERY_DATA 2694 // when changing the audio output device, call addBatteryData to notify 2695 // the change 2696 if ((int)mDevice != value) { 2697 uint32_t params = 0; 2698 // check whether speaker is on 2699 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2700 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2701 } 2702 2703 int deviceWithoutSpeaker 2704 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2705 // check if any other device (except speaker) is on 2706 if (value & deviceWithoutSpeaker ) { 2707 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2708 } 2709 2710 if (params != 0) { 2711 addBatteryData(params); 2712 } 2713 } 2714#endif 2715 2716 // forward device change to effects that have requested to be 2717 // aware of attached audio device. 2718 mDevice = (uint32_t)value; 2719 for (size_t i = 0; i < mEffectChains.size(); i++) { 2720 mEffectChains[i]->setDevice_l(mDevice); 2721 } 2722 } 2723 2724 if (status == NO_ERROR) { 2725 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2726 keyValuePair.string()); 2727 if (!mStandby && status == INVALID_OPERATION) { 2728 mOutput->stream->common.standby(&mOutput->stream->common); 2729 mStandby = true; 2730 mBytesWritten = 0; 2731 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2732 keyValuePair.string()); 2733 } 2734 if (status == NO_ERROR && reconfig) { 2735 delete mAudioMixer; 2736 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2737 mAudioMixer = NULL; 2738 readOutputParameters(); 2739 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2740 for (size_t i = 0; i < mTracks.size() ; i++) { 2741 int name = getTrackName_l(); 2742 if (name < 0) break; 2743 mTracks[i]->mName = name; 2744 // limit track sample rate to 2 x new output sample rate 2745 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2746 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2747 } 2748 } 2749 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2750 } 2751 } 2752 2753 mNewParameters.removeAt(0); 2754 2755 mParamStatus = status; 2756 mParamCond.signal(); 2757 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2758 // already timed out waiting for the status and will never signal the condition. 2759 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2760 } 2761 return reconfig; 2762} 2763 2764status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2765{ 2766 const size_t SIZE = 256; 2767 char buffer[SIZE]; 2768 String8 result; 2769 2770 PlaybackThread::dumpInternals(fd, args); 2771 2772 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2773 result.append(buffer); 2774 write(fd, result.string(), result.size()); 2775 return NO_ERROR; 2776} 2777 2778uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 2779{ 2780 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2781} 2782 2783uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 2784{ 2785 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2786} 2787 2788void AudioFlinger::MixerThread::cacheParameters_l() 2789{ 2790 PlaybackThread::cacheParameters_l(); 2791 2792 // FIXME: Relaxed timing because of a certain device that can't meet latency 2793 // Should be reduced to 2x after the vendor fixes the driver issue 2794 // increase threshold again due to low power audio mode. The way this warning 2795 // threshold is calculated and its usefulness should be reconsidered anyway. 2796 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2797} 2798 2799// ---------------------------------------------------------------------------- 2800AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2801 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2802 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2803 // mLeftVolFloat, mRightVolFloat 2804 // mLeftVolShort, mRightVolShort 2805{ 2806} 2807 2808AudioFlinger::DirectOutputThread::~DirectOutputThread() 2809{ 2810} 2811 2812AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2813 Vector< sp<Track> > *tracksToRemove 2814) 2815{ 2816 sp<Track> trackToRemove; 2817 2818 mixer_state mixerStatus = MIXER_IDLE; 2819 2820 // find out which tracks need to be processed 2821 if (mActiveTracks.size() != 0) { 2822 sp<Track> t = mActiveTracks[0].promote(); 2823 // The track died recently 2824 if (t == 0) return MIXER_IDLE; 2825 2826 Track* const track = t.get(); 2827 audio_track_cblk_t* cblk = track->cblk(); 2828 2829 // The first time a track is added we wait 2830 // for all its buffers to be filled before processing it 2831 if (cblk->framesReady() && track->isReady() && 2832 !track->isPaused() && !track->isTerminated()) 2833 { 2834 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2835 2836 if (track->mFillingUpStatus == Track::FS_FILLED) { 2837 track->mFillingUpStatus = Track::FS_ACTIVE; 2838 mLeftVolFloat = mRightVolFloat = 0; 2839 mLeftVolShort = mRightVolShort = 0; 2840 if (track->mState == TrackBase::RESUMING) { 2841 track->mState = TrackBase::ACTIVE; 2842 rampVolume = true; 2843 } 2844 } else if (cblk->server != 0) { 2845 // If the track is stopped before the first frame was mixed, 2846 // do not apply ramp 2847 rampVolume = true; 2848 } 2849 // compute volume for this track 2850 float left, right; 2851 if (track->isMuted() || mMasterMute || track->isPausing() || 2852 mStreamTypes[track->streamType()].mute) { 2853 left = right = 0; 2854 if (track->isPausing()) { 2855 track->setPaused(); 2856 } 2857 } else { 2858 float typeVolume = mStreamTypes[track->streamType()].volume; 2859 float v = mMasterVolume * typeVolume; 2860 uint32_t vlr = cblk->getVolumeLR(); 2861 float v_clamped = v * (vlr & 0xFFFF); 2862 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2863 left = v_clamped/MAX_GAIN; 2864 v_clamped = v * (vlr >> 16); 2865 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2866 right = v_clamped/MAX_GAIN; 2867 } 2868 2869 if (left != mLeftVolFloat || right != mRightVolFloat) { 2870 mLeftVolFloat = left; 2871 mRightVolFloat = right; 2872 2873 // If audio HAL implements volume control, 2874 // force software volume to nominal value 2875 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2876 left = 1.0f; 2877 right = 1.0f; 2878 } 2879 2880 // Convert volumes from float to 8.24 2881 uint32_t vl = (uint32_t)(left * (1 << 24)); 2882 uint32_t vr = (uint32_t)(right * (1 << 24)); 2883 2884 // Delegate volume control to effect in track effect chain if needed 2885 // only one effect chain can be present on DirectOutputThread, so if 2886 // there is one, the track is connected to it 2887 if (!mEffectChains.isEmpty()) { 2888 // Do not ramp volume if volume is controlled by effect 2889 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2890 rampVolume = false; 2891 } 2892 } 2893 2894 // Convert volumes from 8.24 to 4.12 format 2895 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2896 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2897 leftVol = (uint16_t)v_clamped; 2898 v_clamped = (vr + (1 << 11)) >> 12; 2899 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2900 rightVol = (uint16_t)v_clamped; 2901 } else { 2902 leftVol = mLeftVolShort; 2903 rightVol = mRightVolShort; 2904 rampVolume = false; 2905 } 2906 2907 // reset retry count 2908 track->mRetryCount = kMaxTrackRetriesDirect; 2909 mActiveTrack = t; 2910 mixerStatus = MIXER_TRACKS_READY; 2911 } else { 2912 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2913 if (track->isStopped()) { 2914 track->reset(); 2915 } 2916 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2917 // We have consumed all the buffers of this track. 2918 // Remove it from the list of active tracks. 2919 // TODO: implement behavior for compressed audio 2920 size_t audioHALFrames = 2921 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2922 size_t framesWritten = 2923 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2924 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2925 trackToRemove = track; 2926 } 2927 } else { 2928 // No buffers for this track. Give it a few chances to 2929 // fill a buffer, then remove it from active list. 2930 if (--(track->mRetryCount) <= 0) { 2931 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2932 trackToRemove = track; 2933 } else { 2934 mixerStatus = MIXER_TRACKS_ENABLED; 2935 } 2936 } 2937 } 2938 } 2939 2940 // FIXME merge this with similar code for removing multiple tracks 2941 // remove all the tracks that need to be... 2942 if (CC_UNLIKELY(trackToRemove != 0)) { 2943 tracksToRemove->add(trackToRemove); 2944 mActiveTracks.remove(trackToRemove); 2945 if (!mEffectChains.isEmpty()) { 2946 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2947 trackToRemove->sessionId()); 2948 mEffectChains[0]->decActiveTrackCnt(); 2949 } 2950 if (trackToRemove->isTerminated()) { 2951 removeTrack_l(trackToRemove); 2952 } 2953 } 2954 2955 return mixerStatus; 2956} 2957 2958void AudioFlinger::DirectOutputThread::threadLoop_mix() 2959{ 2960 AudioBufferProvider::Buffer buffer; 2961 size_t frameCount = mFrameCount; 2962 int8_t *curBuf = (int8_t *)mMixBuffer; 2963 // output audio to hardware 2964 while (frameCount) { 2965 buffer.frameCount = frameCount; 2966 mActiveTrack->getNextBuffer(&buffer); 2967 if (CC_UNLIKELY(buffer.raw == NULL)) { 2968 memset(curBuf, 0, frameCount * mFrameSize); 2969 break; 2970 } 2971 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2972 frameCount -= buffer.frameCount; 2973 curBuf += buffer.frameCount * mFrameSize; 2974 mActiveTrack->releaseBuffer(&buffer); 2975 } 2976 sleepTime = 0; 2977 standbyTime = systemTime() + standbyDelay; 2978 mActiveTrack.clear(); 2979 2980 // apply volume 2981 2982 // Do not apply volume on compressed audio 2983 if (!audio_is_linear_pcm(mFormat)) { 2984 return; 2985 } 2986 2987 // convert to signed 16 bit before volume calculation 2988 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2989 size_t count = mFrameCount * mChannelCount; 2990 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2991 int16_t *dst = mMixBuffer + count-1; 2992 while (count--) { 2993 *dst-- = (int16_t)(*src--^0x80) << 8; 2994 } 2995 } 2996 2997 frameCount = mFrameCount; 2998 int16_t *out = mMixBuffer; 2999 if (rampVolume) { 3000 if (mChannelCount == 1) { 3001 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3002 int32_t vlInc = d / (int32_t)frameCount; 3003 int32_t vl = ((int32_t)mLeftVolShort << 16); 3004 do { 3005 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3006 out++; 3007 vl += vlInc; 3008 } while (--frameCount); 3009 3010 } else { 3011 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3012 int32_t vlInc = d / (int32_t)frameCount; 3013 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3014 int32_t vrInc = d / (int32_t)frameCount; 3015 int32_t vl = ((int32_t)mLeftVolShort << 16); 3016 int32_t vr = ((int32_t)mRightVolShort << 16); 3017 do { 3018 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3019 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3020 out += 2; 3021 vl += vlInc; 3022 vr += vrInc; 3023 } while (--frameCount); 3024 } 3025 } else { 3026 if (mChannelCount == 1) { 3027 do { 3028 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3029 out++; 3030 } while (--frameCount); 3031 } else { 3032 do { 3033 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3034 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3035 out += 2; 3036 } while (--frameCount); 3037 } 3038 } 3039 3040 // convert back to unsigned 8 bit after volume calculation 3041 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3042 size_t count = mFrameCount * mChannelCount; 3043 int16_t *src = mMixBuffer; 3044 uint8_t *dst = (uint8_t *)mMixBuffer; 3045 while (count--) { 3046 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3047 } 3048 } 3049 3050 mLeftVolShort = leftVol; 3051 mRightVolShort = rightVol; 3052} 3053 3054void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3055{ 3056 if (sleepTime == 0) { 3057 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3058 sleepTime = activeSleepTime; 3059 } else { 3060 sleepTime = idleSleepTime; 3061 } 3062 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3063 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3064 sleepTime = 0; 3065 } 3066} 3067 3068// getTrackName_l() must be called with ThreadBase::mLock held 3069int AudioFlinger::DirectOutputThread::getTrackName_l() 3070{ 3071 return 0; 3072} 3073 3074// deleteTrackName_l() must be called with ThreadBase::mLock held 3075void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3076{ 3077} 3078 3079// checkForNewParameters_l() must be called with ThreadBase::mLock held 3080bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3081{ 3082 bool reconfig = false; 3083 3084 while (!mNewParameters.isEmpty()) { 3085 status_t status = NO_ERROR; 3086 String8 keyValuePair = mNewParameters[0]; 3087 AudioParameter param = AudioParameter(keyValuePair); 3088 int value; 3089 3090 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3091 // do not accept frame count changes if tracks are open as the track buffer 3092 // size depends on frame count and correct behavior would not be garantied 3093 // if frame count is changed after track creation 3094 if (!mTracks.isEmpty()) { 3095 status = INVALID_OPERATION; 3096 } else { 3097 reconfig = true; 3098 } 3099 } 3100 if (status == NO_ERROR) { 3101 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3102 keyValuePair.string()); 3103 if (!mStandby && status == INVALID_OPERATION) { 3104 mOutput->stream->common.standby(&mOutput->stream->common); 3105 mStandby = true; 3106 mBytesWritten = 0; 3107 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3108 keyValuePair.string()); 3109 } 3110 if (status == NO_ERROR && reconfig) { 3111 readOutputParameters(); 3112 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3113 } 3114 } 3115 3116 mNewParameters.removeAt(0); 3117 3118 mParamStatus = status; 3119 mParamCond.signal(); 3120 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3121 // already timed out waiting for the status and will never signal the condition. 3122 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3123 } 3124 return reconfig; 3125} 3126 3127uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3128{ 3129 uint32_t time; 3130 if (audio_is_linear_pcm(mFormat)) { 3131 time = PlaybackThread::activeSleepTimeUs(); 3132 } else { 3133 time = 10000; 3134 } 3135 return time; 3136} 3137 3138uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3139{ 3140 uint32_t time; 3141 if (audio_is_linear_pcm(mFormat)) { 3142 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3143 } else { 3144 time = 10000; 3145 } 3146 return time; 3147} 3148 3149uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3150{ 3151 uint32_t time; 3152 if (audio_is_linear_pcm(mFormat)) { 3153 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3154 } else { 3155 time = 10000; 3156 } 3157 return time; 3158} 3159 3160void AudioFlinger::DirectOutputThread::cacheParameters_l() 3161{ 3162 PlaybackThread::cacheParameters_l(); 3163 3164 // use shorter standby delay as on normal output to release 3165 // hardware resources as soon as possible 3166 standbyDelay = microseconds(activeSleepTime*2); 3167} 3168 3169// ---------------------------------------------------------------------------- 3170 3171AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3172 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3173 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3174 mWaitTimeMs(UINT_MAX) 3175{ 3176 addOutputTrack(mainThread); 3177} 3178 3179AudioFlinger::DuplicatingThread::~DuplicatingThread() 3180{ 3181 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3182 mOutputTracks[i]->destroy(); 3183 } 3184} 3185 3186void AudioFlinger::DuplicatingThread::threadLoop_mix() 3187{ 3188 // mix buffers... 3189 if (outputsReady(outputTracks)) { 3190 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3191 } else { 3192 memset(mMixBuffer, 0, mixBufferSize); 3193 } 3194 sleepTime = 0; 3195 writeFrames = mFrameCount; 3196} 3197 3198void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3199{ 3200 if (sleepTime == 0) { 3201 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3202 sleepTime = activeSleepTime; 3203 } else { 3204 sleepTime = idleSleepTime; 3205 } 3206 } else if (mBytesWritten != 0) { 3207 // flush remaining overflow buffers in output tracks 3208 for (size_t i = 0; i < outputTracks.size(); i++) { 3209 if (outputTracks[i]->isActive()) { 3210 sleepTime = 0; 3211 writeFrames = 0; 3212 memset(mMixBuffer, 0, mixBufferSize); 3213 break; 3214 } 3215 } 3216 } 3217} 3218 3219void AudioFlinger::DuplicatingThread::threadLoop_write() 3220{ 3221 standbyTime = systemTime() + standbyDelay; 3222 for (size_t i = 0; i < outputTracks.size(); i++) { 3223 outputTracks[i]->write(mMixBuffer, writeFrames); 3224 } 3225 mBytesWritten += mixBufferSize; 3226} 3227 3228void AudioFlinger::DuplicatingThread::threadLoop_standby() 3229{ 3230 // DuplicatingThread implements standby by stopping all tracks 3231 for (size_t i = 0; i < outputTracks.size(); i++) { 3232 outputTracks[i]->stop(); 3233 } 3234} 3235 3236void AudioFlinger::DuplicatingThread::saveOutputTracks() 3237{ 3238 outputTracks = mOutputTracks; 3239} 3240 3241void AudioFlinger::DuplicatingThread::clearOutputTracks() 3242{ 3243 outputTracks.clear(); 3244} 3245 3246void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3247{ 3248 Mutex::Autolock _l(mLock); 3249 // FIXME explain this formula 3250 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3251 OutputTrack *outputTrack = new OutputTrack(thread, 3252 this, 3253 mSampleRate, 3254 mFormat, 3255 mChannelMask, 3256 frameCount); 3257 if (outputTrack->cblk() != NULL) { 3258 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3259 mOutputTracks.add(outputTrack); 3260 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3261 updateWaitTime_l(); 3262 } 3263} 3264 3265void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3266{ 3267 Mutex::Autolock _l(mLock); 3268 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3269 if (mOutputTracks[i]->thread() == thread) { 3270 mOutputTracks[i]->destroy(); 3271 mOutputTracks.removeAt(i); 3272 updateWaitTime_l(); 3273 return; 3274 } 3275 } 3276 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3277} 3278 3279// caller must hold mLock 3280void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3281{ 3282 mWaitTimeMs = UINT_MAX; 3283 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3284 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3285 if (strong != 0) { 3286 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3287 if (waitTimeMs < mWaitTimeMs) { 3288 mWaitTimeMs = waitTimeMs; 3289 } 3290 } 3291 } 3292} 3293 3294 3295bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3296{ 3297 for (size_t i = 0; i < outputTracks.size(); i++) { 3298 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3299 if (thread == 0) { 3300 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3301 return false; 3302 } 3303 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3304 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3305 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3306 return false; 3307 } 3308 } 3309 return true; 3310} 3311 3312uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3313{ 3314 return (mWaitTimeMs * 1000) / 2; 3315} 3316 3317void AudioFlinger::DuplicatingThread::cacheParameters_l() 3318{ 3319 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3320 updateWaitTime_l(); 3321 3322 MixerThread::cacheParameters_l(); 3323} 3324 3325// ---------------------------------------------------------------------------- 3326 3327// TrackBase constructor must be called with AudioFlinger::mLock held 3328AudioFlinger::ThreadBase::TrackBase::TrackBase( 3329 ThreadBase *thread, 3330 const sp<Client>& client, 3331 uint32_t sampleRate, 3332 audio_format_t format, 3333 uint32_t channelMask, 3334 int frameCount, 3335 const sp<IMemory>& sharedBuffer, 3336 int sessionId) 3337 : RefBase(), 3338 mThread(thread), 3339 mClient(client), 3340 mCblk(NULL), 3341 // mBuffer 3342 // mBufferEnd 3343 mFrameCount(0), 3344 mState(IDLE), 3345 mFormat(format), 3346 mStepServerFailed(false), 3347 mSessionId(sessionId) 3348 // mChannelCount 3349 // mChannelMask 3350{ 3351 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3352 3353 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3354 size_t size = sizeof(audio_track_cblk_t); 3355 uint8_t channelCount = popcount(channelMask); 3356 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3357 if (sharedBuffer == 0) { 3358 size += bufferSize; 3359 } 3360 3361 if (client != NULL) { 3362 mCblkMemory = client->heap()->allocate(size); 3363 if (mCblkMemory != 0) { 3364 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3365 if (mCblk != NULL) { // construct the shared structure in-place. 3366 new(mCblk) audio_track_cblk_t(); 3367 // clear all buffers 3368 mCblk->frameCount = frameCount; 3369 mCblk->sampleRate = sampleRate; 3370// uncomment the following lines to quickly test 32-bit wraparound 3371// mCblk->user = 0xffff0000; 3372// mCblk->server = 0xffff0000; 3373// mCblk->userBase = 0xffff0000; 3374// mCblk->serverBase = 0xffff0000; 3375 mChannelCount = channelCount; 3376 mChannelMask = channelMask; 3377 if (sharedBuffer == 0) { 3378 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3379 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3380 // Force underrun condition to avoid false underrun callback until first data is 3381 // written to buffer (other flags are cleared) 3382 mCblk->flags = CBLK_UNDERRUN_ON; 3383 } else { 3384 mBuffer = sharedBuffer->pointer(); 3385 } 3386 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3387 } 3388 } else { 3389 ALOGE("not enough memory for AudioTrack size=%u", size); 3390 client->heap()->dump("AudioTrack"); 3391 return; 3392 } 3393 } else { 3394 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3395 // construct the shared structure in-place. 3396 new(mCblk) audio_track_cblk_t(); 3397 // clear all buffers 3398 mCblk->frameCount = frameCount; 3399 mCblk->sampleRate = sampleRate; 3400// uncomment the following lines to quickly test 32-bit wraparound 3401// mCblk->user = 0xffff0000; 3402// mCblk->server = 0xffff0000; 3403// mCblk->userBase = 0xffff0000; 3404// mCblk->serverBase = 0xffff0000; 3405 mChannelCount = channelCount; 3406 mChannelMask = channelMask; 3407 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3408 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3409 // Force underrun condition to avoid false underrun callback until first data is 3410 // written to buffer (other flags are cleared) 3411 mCblk->flags = CBLK_UNDERRUN_ON; 3412 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3413 } 3414} 3415 3416AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3417{ 3418 if (mCblk != NULL) { 3419 if (mClient == 0) { 3420 delete mCblk; 3421 } else { 3422 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3423 } 3424 } 3425 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3426 if (mClient != 0) { 3427 // Client destructor must run with AudioFlinger mutex locked 3428 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3429 // If the client's reference count drops to zero, the associated destructor 3430 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3431 // relying on the automatic clear() at end of scope. 3432 mClient.clear(); 3433 } 3434} 3435 3436// AudioBufferProvider interface 3437// getNextBuffer() = 0; 3438// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3439void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3440{ 3441 buffer->raw = NULL; 3442 mFrameCount = buffer->frameCount; 3443 (void) step(); // ignore return value of step() 3444 buffer->frameCount = 0; 3445} 3446 3447bool AudioFlinger::ThreadBase::TrackBase::step() { 3448 bool result; 3449 audio_track_cblk_t* cblk = this->cblk(); 3450 3451 result = cblk->stepServer(mFrameCount); 3452 if (!result) { 3453 ALOGV("stepServer failed acquiring cblk mutex"); 3454 mStepServerFailed = true; 3455 } 3456 return result; 3457} 3458 3459void AudioFlinger::ThreadBase::TrackBase::reset() { 3460 audio_track_cblk_t* cblk = this->cblk(); 3461 3462 cblk->user = 0; 3463 cblk->server = 0; 3464 cblk->userBase = 0; 3465 cblk->serverBase = 0; 3466 mStepServerFailed = false; 3467 ALOGV("TrackBase::reset"); 3468} 3469 3470int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3471 return (int)mCblk->sampleRate; 3472} 3473 3474void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3475 audio_track_cblk_t* cblk = this->cblk(); 3476 size_t frameSize = cblk->frameSize; 3477 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3478 int8_t *bufferEnd = bufferStart + frames * frameSize; 3479 3480 // Check validity of returned pointer in case the track control block would have been corrupted. 3481 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3482 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3483 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3484 server %u, serverBase %u, user %u, userBase %u", 3485 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3486 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3487 return NULL; 3488 } 3489 3490 return bufferStart; 3491} 3492 3493status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3494{ 3495 mSyncEvents.add(event); 3496 return NO_ERROR; 3497} 3498 3499// ---------------------------------------------------------------------------- 3500 3501// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3502AudioFlinger::PlaybackThread::Track::Track( 3503 PlaybackThread *thread, 3504 const sp<Client>& client, 3505 audio_stream_type_t streamType, 3506 uint32_t sampleRate, 3507 audio_format_t format, 3508 uint32_t channelMask, 3509 int frameCount, 3510 const sp<IMemory>& sharedBuffer, 3511 int sessionId, 3512 IAudioFlinger::track_flags_t flags) 3513 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3514 mMute(false), 3515 // mFillingUpStatus ? 3516 // mRetryCount initialized later when needed 3517 mSharedBuffer(sharedBuffer), 3518 mStreamType(streamType), 3519 mName(-1), // see note below 3520 mMainBuffer(thread->mixBuffer()), 3521 mAuxBuffer(NULL), 3522 mAuxEffectId(0), mHasVolumeController(false), 3523 mPresentationCompleteFrames(0), 3524 mFlags(flags) 3525{ 3526 if (mCblk != NULL) { 3527 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3528 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3529 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3530 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3531 mName = thread->getTrackName_l(); 3532 if (mName < 0) { 3533 ALOGE("no more track names available"); 3534 } 3535 } 3536 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3537} 3538 3539AudioFlinger::PlaybackThread::Track::~Track() 3540{ 3541 ALOGV("PlaybackThread::Track destructor"); 3542 sp<ThreadBase> thread = mThread.promote(); 3543 if (thread != 0) { 3544 Mutex::Autolock _l(thread->mLock); 3545 mState = TERMINATED; 3546 } 3547} 3548 3549void AudioFlinger::PlaybackThread::Track::destroy() 3550{ 3551 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3552 // by removing it from mTracks vector, so there is a risk that this Tracks's 3553 // destructor is called. As the destructor needs to lock mLock, 3554 // we must acquire a strong reference on this Track before locking mLock 3555 // here so that the destructor is called only when exiting this function. 3556 // On the other hand, as long as Track::destroy() is only called by 3557 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3558 // this Track with its member mTrack. 3559 sp<Track> keep(this); 3560 { // scope for mLock 3561 sp<ThreadBase> thread = mThread.promote(); 3562 if (thread != 0) { 3563 if (!isOutputTrack()) { 3564 if (mState == ACTIVE || mState == RESUMING) { 3565 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3566 3567#ifdef ADD_BATTERY_DATA 3568 // to track the speaker usage 3569 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3570#endif 3571 } 3572 AudioSystem::releaseOutput(thread->id()); 3573 } 3574 Mutex::Autolock _l(thread->mLock); 3575 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3576 playbackThread->destroyTrack_l(this); 3577 } 3578 } 3579} 3580 3581void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3582{ 3583 uint32_t vlr = mCblk->getVolumeLR(); 3584 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", 3585 mName - AudioMixer::TRACK0, 3586 (mClient == 0) ? getpid_cached : mClient->pid(), 3587 mStreamType, 3588 mFormat, 3589 mChannelMask, 3590 mSessionId, 3591 mFrameCount, 3592 mState, 3593 mMute, 3594 mFillingUpStatus, 3595 mCblk->sampleRate, 3596 vlr & 0xFFFF, 3597 vlr >> 16, 3598 mCblk->server, 3599 mCblk->user, 3600 (int)mMainBuffer, 3601 (int)mAuxBuffer); 3602} 3603 3604// AudioBufferProvider interface 3605status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3606 AudioBufferProvider::Buffer* buffer, int64_t pts) 3607{ 3608 audio_track_cblk_t* cblk = this->cblk(); 3609 uint32_t framesReady; 3610 uint32_t framesReq = buffer->frameCount; 3611 3612 // Check if last stepServer failed, try to step now 3613 if (mStepServerFailed) { 3614 if (!step()) goto getNextBuffer_exit; 3615 ALOGV("stepServer recovered"); 3616 mStepServerFailed = false; 3617 } 3618 3619 framesReady = cblk->framesReady(); 3620 3621 if (CC_LIKELY(framesReady)) { 3622 uint32_t s = cblk->server; 3623 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3624 3625 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3626 if (framesReq > framesReady) { 3627 framesReq = framesReady; 3628 } 3629 if (framesReq > bufferEnd - s) { 3630 framesReq = bufferEnd - s; 3631 } 3632 3633 buffer->raw = getBuffer(s, framesReq); 3634 if (buffer->raw == NULL) goto getNextBuffer_exit; 3635 3636 buffer->frameCount = framesReq; 3637 return NO_ERROR; 3638 } 3639 3640getNextBuffer_exit: 3641 buffer->raw = NULL; 3642 buffer->frameCount = 0; 3643 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3644 return NOT_ENOUGH_DATA; 3645} 3646 3647uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3648 return mCblk->framesReady(); 3649} 3650 3651bool AudioFlinger::PlaybackThread::Track::isReady() const { 3652 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3653 3654 if (framesReady() >= mCblk->frameCount || 3655 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3656 mFillingUpStatus = FS_FILLED; 3657 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3658 return true; 3659 } 3660 return false; 3661} 3662 3663status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid, 3664 AudioSystem::sync_event_t event, 3665 int triggerSession) 3666{ 3667 status_t status = NO_ERROR; 3668 ALOGV("start(%d), calling pid %d session %d tid %d", 3669 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3670 // check for use case 2 with missing callback 3671 if (isFastTrack() && (mSharedBuffer == 0) && (tid == 0)) { 3672 ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied"); 3673 mFlags &= ~IAudioFlinger::TRACK_FAST; 3674 // FIXME the track must be invalidated and moved to another thread or 3675 // attached directly to the normal mixer now 3676 } 3677 sp<ThreadBase> thread = mThread.promote(); 3678 if (thread != 0) { 3679 Mutex::Autolock _l(thread->mLock); 3680 track_state state = mState; 3681 // here the track could be either new, or restarted 3682 // in both cases "unstop" the track 3683 if (mState == PAUSED) { 3684 mState = TrackBase::RESUMING; 3685 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3686 } else { 3687 mState = TrackBase::ACTIVE; 3688 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3689 } 3690 3691 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3692 thread->mLock.unlock(); 3693 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3694 thread->mLock.lock(); 3695 3696#ifdef ADD_BATTERY_DATA 3697 // to track the speaker usage 3698 if (status == NO_ERROR) { 3699 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3700 } 3701#endif 3702 } 3703 if (status == NO_ERROR) { 3704 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3705 playbackThread->addTrack_l(this); 3706 } else { 3707 mState = state; 3708 } 3709 } else { 3710 status = BAD_VALUE; 3711 } 3712 return status; 3713} 3714 3715void AudioFlinger::PlaybackThread::Track::stop() 3716{ 3717 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3718 sp<ThreadBase> thread = mThread.promote(); 3719 if (thread != 0) { 3720 Mutex::Autolock _l(thread->mLock); 3721 track_state state = mState; 3722 if (mState > STOPPED) { 3723 mState = STOPPED; 3724 // If the track is not active (PAUSED and buffers full), flush buffers 3725 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3726 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3727 reset(); 3728 } 3729 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3730 } 3731 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3732 thread->mLock.unlock(); 3733 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3734 thread->mLock.lock(); 3735 3736#ifdef ADD_BATTERY_DATA 3737 // to track the speaker usage 3738 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3739#endif 3740 } 3741 } 3742} 3743 3744void AudioFlinger::PlaybackThread::Track::pause() 3745{ 3746 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3747 sp<ThreadBase> thread = mThread.promote(); 3748 if (thread != 0) { 3749 Mutex::Autolock _l(thread->mLock); 3750 if (mState == ACTIVE || mState == RESUMING) { 3751 mState = PAUSING; 3752 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3753 if (!isOutputTrack()) { 3754 thread->mLock.unlock(); 3755 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3756 thread->mLock.lock(); 3757 3758#ifdef ADD_BATTERY_DATA 3759 // to track the speaker usage 3760 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3761#endif 3762 } 3763 } 3764 } 3765} 3766 3767void AudioFlinger::PlaybackThread::Track::flush() 3768{ 3769 ALOGV("flush(%d)", mName); 3770 sp<ThreadBase> thread = mThread.promote(); 3771 if (thread != 0) { 3772 Mutex::Autolock _l(thread->mLock); 3773 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3774 return; 3775 } 3776 // No point remaining in PAUSED state after a flush => go to 3777 // STOPPED state 3778 mState = STOPPED; 3779 3780 // do not reset the track if it is still in the process of being stopped or paused. 3781 // this will be done by prepareTracks_l() when the track is stopped. 3782 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3783 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3784 reset(); 3785 } 3786 } 3787} 3788 3789void AudioFlinger::PlaybackThread::Track::reset() 3790{ 3791 // Do not reset twice to avoid discarding data written just after a flush and before 3792 // the audioflinger thread detects the track is stopped. 3793 if (!mResetDone) { 3794 TrackBase::reset(); 3795 // Force underrun condition to avoid false underrun callback until first data is 3796 // written to buffer 3797 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3798 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3799 mFillingUpStatus = FS_FILLING; 3800 mResetDone = true; 3801 mPresentationCompleteFrames = 0; 3802 } 3803} 3804 3805void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3806{ 3807 mMute = muted; 3808} 3809 3810status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3811{ 3812 status_t status = DEAD_OBJECT; 3813 sp<ThreadBase> thread = mThread.promote(); 3814 if (thread != 0) { 3815 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3816 status = playbackThread->attachAuxEffect(this, EffectId); 3817 } 3818 return status; 3819} 3820 3821void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3822{ 3823 mAuxEffectId = EffectId; 3824 mAuxBuffer = buffer; 3825} 3826 3827bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 3828 size_t audioHalFrames) 3829{ 3830 // a track is considered presented when the total number of frames written to audio HAL 3831 // corresponds to the number of frames written when presentationComplete() is called for the 3832 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 3833 if (mPresentationCompleteFrames == 0) { 3834 mPresentationCompleteFrames = framesWritten + audioHalFrames; 3835 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 3836 mPresentationCompleteFrames, audioHalFrames); 3837 } 3838 if (framesWritten >= mPresentationCompleteFrames) { 3839 ALOGV("presentationComplete() session %d complete: framesWritten %d", 3840 mSessionId, framesWritten); 3841 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 3842 mPresentationCompleteFrames = 0; 3843 return true; 3844 } 3845 return false; 3846} 3847 3848void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 3849{ 3850 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 3851 if (mSyncEvents[i]->type() == type) { 3852 mSyncEvents[i]->trigger(); 3853 mSyncEvents.removeAt(i); 3854 i--; 3855 } 3856 } 3857} 3858 3859 3860// timed audio tracks 3861 3862sp<AudioFlinger::PlaybackThread::TimedTrack> 3863AudioFlinger::PlaybackThread::TimedTrack::create( 3864 PlaybackThread *thread, 3865 const sp<Client>& client, 3866 audio_stream_type_t streamType, 3867 uint32_t sampleRate, 3868 audio_format_t format, 3869 uint32_t channelMask, 3870 int frameCount, 3871 const sp<IMemory>& sharedBuffer, 3872 int sessionId) { 3873 if (!client->reserveTimedTrack()) 3874 return NULL; 3875 3876 return new TimedTrack( 3877 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3878 sharedBuffer, sessionId); 3879} 3880 3881AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3882 PlaybackThread *thread, 3883 const sp<Client>& client, 3884 audio_stream_type_t streamType, 3885 uint32_t sampleRate, 3886 audio_format_t format, 3887 uint32_t channelMask, 3888 int frameCount, 3889 const sp<IMemory>& sharedBuffer, 3890 int sessionId) 3891 : Track(thread, client, streamType, sampleRate, format, channelMask, 3892 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 3893 mTimedSilenceBuffer(NULL), 3894 mTimedSilenceBufferSize(0), 3895 mTimedAudioOutputOnTime(false), 3896 mMediaTimeTransformValid(false) 3897{ 3898 LocalClock lc; 3899 mLocalTimeFreq = lc.getLocalFreq(); 3900 3901 mLocalTimeToSampleTransform.a_zero = 0; 3902 mLocalTimeToSampleTransform.b_zero = 0; 3903 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3904 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3905 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3906 &mLocalTimeToSampleTransform.a_to_b_denom); 3907} 3908 3909AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3910 mClient->releaseTimedTrack(); 3911 delete [] mTimedSilenceBuffer; 3912} 3913 3914status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3915 size_t size, sp<IMemory>* buffer) { 3916 3917 Mutex::Autolock _l(mTimedBufferQueueLock); 3918 3919 trimTimedBufferQueue_l(); 3920 3921 // lazily initialize the shared memory heap for timed buffers 3922 if (mTimedMemoryDealer == NULL) { 3923 const int kTimedBufferHeapSize = 512 << 10; 3924 3925 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3926 "AudioFlingerTimed"); 3927 if (mTimedMemoryDealer == NULL) 3928 return NO_MEMORY; 3929 } 3930 3931 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3932 if (newBuffer == NULL) { 3933 newBuffer = mTimedMemoryDealer->allocate(size); 3934 if (newBuffer == NULL) 3935 return NO_MEMORY; 3936 } 3937 3938 *buffer = newBuffer; 3939 return NO_ERROR; 3940} 3941 3942// caller must hold mTimedBufferQueueLock 3943void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3944 int64_t mediaTimeNow; 3945 { 3946 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3947 if (!mMediaTimeTransformValid) 3948 return; 3949 3950 int64_t targetTimeNow; 3951 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3952 ? mCCHelper.getCommonTime(&targetTimeNow) 3953 : mCCHelper.getLocalTime(&targetTimeNow); 3954 3955 if (OK != res) 3956 return; 3957 3958 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3959 &mediaTimeNow)) { 3960 return; 3961 } 3962 } 3963 3964 size_t trimIndex; 3965 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3966 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3967 break; 3968 } 3969 3970 if (trimIndex) { 3971 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3972 } 3973} 3974 3975status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3976 const sp<IMemory>& buffer, int64_t pts) { 3977 3978 { 3979 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3980 if (!mMediaTimeTransformValid) 3981 return INVALID_OPERATION; 3982 } 3983 3984 Mutex::Autolock _l(mTimedBufferQueueLock); 3985 3986 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3987 3988 return NO_ERROR; 3989} 3990 3991status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3992 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3993 3994 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3995 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3996 target); 3997 3998 if (!(target == TimedAudioTrack::LOCAL_TIME || 3999 target == TimedAudioTrack::COMMON_TIME)) { 4000 return BAD_VALUE; 4001 } 4002 4003 Mutex::Autolock lock(mMediaTimeTransformLock); 4004 mMediaTimeTransform = xform; 4005 mMediaTimeTransformTarget = target; 4006 mMediaTimeTransformValid = true; 4007 4008 return NO_ERROR; 4009} 4010 4011#define min(a, b) ((a) < (b) ? (a) : (b)) 4012 4013// implementation of getNextBuffer for tracks whose buffers have timestamps 4014status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4015 AudioBufferProvider::Buffer* buffer, int64_t pts) 4016{ 4017 if (pts == AudioBufferProvider::kInvalidPTS) { 4018 buffer->raw = 0; 4019 buffer->frameCount = 0; 4020 return INVALID_OPERATION; 4021 } 4022 4023 Mutex::Autolock _l(mTimedBufferQueueLock); 4024 4025 while (true) { 4026 4027 // if we have no timed buffers, then fail 4028 if (mTimedBufferQueue.isEmpty()) { 4029 buffer->raw = 0; 4030 buffer->frameCount = 0; 4031 return NOT_ENOUGH_DATA; 4032 } 4033 4034 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4035 4036 // calculate the PTS of the head of the timed buffer queue expressed in 4037 // local time 4038 int64_t headLocalPTS; 4039 { 4040 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4041 4042 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4043 4044 if (mMediaTimeTransform.a_to_b_denom == 0) { 4045 // the transform represents a pause, so yield silence 4046 timedYieldSilence(buffer->frameCount, buffer); 4047 return NO_ERROR; 4048 } 4049 4050 int64_t transformedPTS; 4051 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4052 &transformedPTS)) { 4053 // the transform failed. this shouldn't happen, but if it does 4054 // then just drop this buffer 4055 ALOGW("timedGetNextBuffer transform failed"); 4056 buffer->raw = 0; 4057 buffer->frameCount = 0; 4058 mTimedBufferQueue.removeAt(0); 4059 return NO_ERROR; 4060 } 4061 4062 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4063 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4064 &headLocalPTS)) { 4065 buffer->raw = 0; 4066 buffer->frameCount = 0; 4067 return INVALID_OPERATION; 4068 } 4069 } else { 4070 headLocalPTS = transformedPTS; 4071 } 4072 } 4073 4074 // adjust the head buffer's PTS to reflect the portion of the head buffer 4075 // that has already been consumed 4076 int64_t effectivePTS = headLocalPTS + 4077 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4078 4079 // Calculate the delta in samples between the head of the input buffer 4080 // queue and the start of the next output buffer that will be written. 4081 // If the transformation fails because of over or underflow, it means 4082 // that the sample's position in the output stream is so far out of 4083 // whack that it should just be dropped. 4084 int64_t sampleDelta; 4085 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4086 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4087 mTimedBufferQueue.removeAt(0); 4088 continue; 4089 } 4090 if (!mLocalTimeToSampleTransform.doForwardTransform( 4091 (effectivePTS - pts) << 32, &sampleDelta)) { 4092 ALOGV("*** too late during sample rate transform: dropped buffer"); 4093 mTimedBufferQueue.removeAt(0); 4094 continue; 4095 } 4096 4097 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4098 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4099 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4100 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4101 4102 // if the delta between the ideal placement for the next input sample and 4103 // the current output position is within this threshold, then we will 4104 // concatenate the next input samples to the previous output 4105 const int64_t kSampleContinuityThreshold = 4106 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4107 4108 // if this is the first buffer of audio that we're emitting from this track 4109 // then it should be almost exactly on time. 4110 const int64_t kSampleStartupThreshold = 1LL << 32; 4111 4112 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4113 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4114 // the next input is close enough to being on time, so concatenate it 4115 // with the last output 4116 timedYieldSamples(buffer); 4117 4118 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4119 return NO_ERROR; 4120 } else if (sampleDelta > 0) { 4121 // the gap between the current output position and the proper start of 4122 // the next input sample is too big, so fill it with silence 4123 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4124 4125 timedYieldSilence(framesUntilNextInput, buffer); 4126 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4127 return NO_ERROR; 4128 } else { 4129 // the next input sample is late 4130 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4131 size_t onTimeSamplePosition = 4132 head.position() + lateFrames * mCblk->frameSize; 4133 4134 if (onTimeSamplePosition > head.buffer()->size()) { 4135 // all the remaining samples in the head are too late, so 4136 // drop it and move on 4137 ALOGV("*** too late: dropped buffer"); 4138 mTimedBufferQueue.removeAt(0); 4139 continue; 4140 } else { 4141 // skip over the late samples 4142 head.setPosition(onTimeSamplePosition); 4143 4144 // yield the available samples 4145 timedYieldSamples(buffer); 4146 4147 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4148 return NO_ERROR; 4149 } 4150 } 4151 } 4152} 4153 4154// Yield samples from the timed buffer queue head up to the given output 4155// buffer's capacity. 4156// 4157// Caller must hold mTimedBufferQueueLock 4158void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4159 AudioBufferProvider::Buffer* buffer) { 4160 4161 const TimedBuffer& head = mTimedBufferQueue[0]; 4162 4163 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4164 head.position()); 4165 4166 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4167 mCblk->frameSize); 4168 size_t framesRequested = buffer->frameCount; 4169 buffer->frameCount = min(framesLeftInHead, framesRequested); 4170 4171 mTimedAudioOutputOnTime = true; 4172} 4173 4174// Yield samples of silence up to the given output buffer's capacity 4175// 4176// Caller must hold mTimedBufferQueueLock 4177void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4178 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4179 4180 // lazily allocate a buffer filled with silence 4181 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4182 delete [] mTimedSilenceBuffer; 4183 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4184 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4185 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4186 } 4187 4188 buffer->raw = mTimedSilenceBuffer; 4189 size_t framesRequested = buffer->frameCount; 4190 buffer->frameCount = min(numFrames, framesRequested); 4191 4192 mTimedAudioOutputOnTime = false; 4193} 4194 4195// AudioBufferProvider interface 4196void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4197 AudioBufferProvider::Buffer* buffer) { 4198 4199 Mutex::Autolock _l(mTimedBufferQueueLock); 4200 4201 // If the buffer which was just released is part of the buffer at the head 4202 // of the queue, be sure to update the amt of the buffer which has been 4203 // consumed. If the buffer being returned is not part of the head of the 4204 // queue, its either because the buffer is part of the silence buffer, or 4205 // because the head of the timed queue was trimmed after the mixer called 4206 // getNextBuffer but before the mixer called releaseBuffer. 4207 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4208 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4209 4210 void* start = head.buffer()->pointer(); 4211 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4212 4213 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4214 head.setPosition(head.position() + 4215 (buffer->frameCount * mCblk->frameSize)); 4216 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4217 mTimedBufferQueue.removeAt(0); 4218 } 4219 } 4220 } 4221 4222 buffer->raw = 0; 4223 buffer->frameCount = 0; 4224} 4225 4226uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4227 Mutex::Autolock _l(mTimedBufferQueueLock); 4228 4229 uint32_t frames = 0; 4230 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4231 const TimedBuffer& tb = mTimedBufferQueue[i]; 4232 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4233 } 4234 4235 return frames; 4236} 4237 4238AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4239 : mPTS(0), mPosition(0) {} 4240 4241AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4242 const sp<IMemory>& buffer, int64_t pts) 4243 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4244 4245// ---------------------------------------------------------------------------- 4246 4247// RecordTrack constructor must be called with AudioFlinger::mLock held 4248AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4249 RecordThread *thread, 4250 const sp<Client>& client, 4251 uint32_t sampleRate, 4252 audio_format_t format, 4253 uint32_t channelMask, 4254 int frameCount, 4255 int sessionId) 4256 : TrackBase(thread, client, sampleRate, format, 4257 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4258 mOverflow(false) 4259{ 4260 if (mCblk != NULL) { 4261 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4262 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4263 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4264 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4265 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4266 } else { 4267 mCblk->frameSize = sizeof(int8_t); 4268 } 4269 } 4270} 4271 4272AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4273{ 4274 sp<ThreadBase> thread = mThread.promote(); 4275 if (thread != 0) { 4276 AudioSystem::releaseInput(thread->id()); 4277 } 4278} 4279 4280// AudioBufferProvider interface 4281status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4282{ 4283 audio_track_cblk_t* cblk = this->cblk(); 4284 uint32_t framesAvail; 4285 uint32_t framesReq = buffer->frameCount; 4286 4287 // Check if last stepServer failed, try to step now 4288 if (mStepServerFailed) { 4289 if (!step()) goto getNextBuffer_exit; 4290 ALOGV("stepServer recovered"); 4291 mStepServerFailed = false; 4292 } 4293 4294 framesAvail = cblk->framesAvailable_l(); 4295 4296 if (CC_LIKELY(framesAvail)) { 4297 uint32_t s = cblk->server; 4298 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4299 4300 if (framesReq > framesAvail) { 4301 framesReq = framesAvail; 4302 } 4303 if (framesReq > bufferEnd - s) { 4304 framesReq = bufferEnd - s; 4305 } 4306 4307 buffer->raw = getBuffer(s, framesReq); 4308 if (buffer->raw == NULL) goto getNextBuffer_exit; 4309 4310 buffer->frameCount = framesReq; 4311 return NO_ERROR; 4312 } 4313 4314getNextBuffer_exit: 4315 buffer->raw = NULL; 4316 buffer->frameCount = 0; 4317 return NOT_ENOUGH_DATA; 4318} 4319 4320status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid, 4321 AudioSystem::sync_event_t event, 4322 int triggerSession) 4323{ 4324 sp<ThreadBase> thread = mThread.promote(); 4325 if (thread != 0) { 4326 RecordThread *recordThread = (RecordThread *)thread.get(); 4327 return recordThread->start(this, tid, event, triggerSession); 4328 } else { 4329 return BAD_VALUE; 4330 } 4331} 4332 4333void AudioFlinger::RecordThread::RecordTrack::stop() 4334{ 4335 sp<ThreadBase> thread = mThread.promote(); 4336 if (thread != 0) { 4337 RecordThread *recordThread = (RecordThread *)thread.get(); 4338 recordThread->stop(this); 4339 TrackBase::reset(); 4340 // Force overrun condition to avoid false overrun callback until first data is 4341 // read from buffer 4342 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4343 } 4344} 4345 4346void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4347{ 4348 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4349 (mClient == 0) ? getpid_cached : mClient->pid(), 4350 mFormat, 4351 mChannelMask, 4352 mSessionId, 4353 mFrameCount, 4354 mState, 4355 mCblk->sampleRate, 4356 mCblk->server, 4357 mCblk->user); 4358} 4359 4360 4361// ---------------------------------------------------------------------------- 4362 4363AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4364 PlaybackThread *playbackThread, 4365 DuplicatingThread *sourceThread, 4366 uint32_t sampleRate, 4367 audio_format_t format, 4368 uint32_t channelMask, 4369 int frameCount) 4370 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 4371 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 4372 mActive(false), mSourceThread(sourceThread) 4373{ 4374 4375 if (mCblk != NULL) { 4376 mCblk->flags |= CBLK_DIRECTION_OUT; 4377 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4378 mOutBuffer.frameCount = 0; 4379 playbackThread->mTracks.add(this); 4380 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4381 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4382 mCblk, mBuffer, mCblk->buffers, 4383 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4384 } else { 4385 ALOGW("Error creating output track on thread %p", playbackThread); 4386 } 4387} 4388 4389AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4390{ 4391 clearBufferQueue(); 4392} 4393 4394status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid, 4395 AudioSystem::sync_event_t event, 4396 int triggerSession) 4397{ 4398 status_t status = Track::start(tid, event, triggerSession); 4399 if (status != NO_ERROR) { 4400 return status; 4401 } 4402 4403 mActive = true; 4404 mRetryCount = 127; 4405 return status; 4406} 4407 4408void AudioFlinger::PlaybackThread::OutputTrack::stop() 4409{ 4410 Track::stop(); 4411 clearBufferQueue(); 4412 mOutBuffer.frameCount = 0; 4413 mActive = false; 4414} 4415 4416bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4417{ 4418 Buffer *pInBuffer; 4419 Buffer inBuffer; 4420 uint32_t channelCount = mChannelCount; 4421 bool outputBufferFull = false; 4422 inBuffer.frameCount = frames; 4423 inBuffer.i16 = data; 4424 4425 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4426 4427 if (!mActive && frames != 0) { 4428 start(0); 4429 sp<ThreadBase> thread = mThread.promote(); 4430 if (thread != 0) { 4431 MixerThread *mixerThread = (MixerThread *)thread.get(); 4432 if (mCblk->frameCount > frames){ 4433 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4434 uint32_t startFrames = (mCblk->frameCount - frames); 4435 pInBuffer = new Buffer; 4436 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4437 pInBuffer->frameCount = startFrames; 4438 pInBuffer->i16 = pInBuffer->mBuffer; 4439 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4440 mBufferQueue.add(pInBuffer); 4441 } else { 4442 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4443 } 4444 } 4445 } 4446 } 4447 4448 while (waitTimeLeftMs) { 4449 // First write pending buffers, then new data 4450 if (mBufferQueue.size()) { 4451 pInBuffer = mBufferQueue.itemAt(0); 4452 } else { 4453 pInBuffer = &inBuffer; 4454 } 4455 4456 if (pInBuffer->frameCount == 0) { 4457 break; 4458 } 4459 4460 if (mOutBuffer.frameCount == 0) { 4461 mOutBuffer.frameCount = pInBuffer->frameCount; 4462 nsecs_t startTime = systemTime(); 4463 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4464 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4465 outputBufferFull = true; 4466 break; 4467 } 4468 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4469 if (waitTimeLeftMs >= waitTimeMs) { 4470 waitTimeLeftMs -= waitTimeMs; 4471 } else { 4472 waitTimeLeftMs = 0; 4473 } 4474 } 4475 4476 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4477 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4478 mCblk->stepUser(outFrames); 4479 pInBuffer->frameCount -= outFrames; 4480 pInBuffer->i16 += outFrames * channelCount; 4481 mOutBuffer.frameCount -= outFrames; 4482 mOutBuffer.i16 += outFrames * channelCount; 4483 4484 if (pInBuffer->frameCount == 0) { 4485 if (mBufferQueue.size()) { 4486 mBufferQueue.removeAt(0); 4487 delete [] pInBuffer->mBuffer; 4488 delete pInBuffer; 4489 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4490 } else { 4491 break; 4492 } 4493 } 4494 } 4495 4496 // If we could not write all frames, allocate a buffer and queue it for next time. 4497 if (inBuffer.frameCount) { 4498 sp<ThreadBase> thread = mThread.promote(); 4499 if (thread != 0 && !thread->standby()) { 4500 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4501 pInBuffer = new Buffer; 4502 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4503 pInBuffer->frameCount = inBuffer.frameCount; 4504 pInBuffer->i16 = pInBuffer->mBuffer; 4505 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4506 mBufferQueue.add(pInBuffer); 4507 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4508 } else { 4509 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4510 } 4511 } 4512 } 4513 4514 // Calling write() with a 0 length buffer, means that no more data will be written: 4515 // If no more buffers are pending, fill output track buffer to make sure it is started 4516 // by output mixer. 4517 if (frames == 0 && mBufferQueue.size() == 0) { 4518 if (mCblk->user < mCblk->frameCount) { 4519 frames = mCblk->frameCount - mCblk->user; 4520 pInBuffer = new Buffer; 4521 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4522 pInBuffer->frameCount = frames; 4523 pInBuffer->i16 = pInBuffer->mBuffer; 4524 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4525 mBufferQueue.add(pInBuffer); 4526 } else if (mActive) { 4527 stop(); 4528 } 4529 } 4530 4531 return outputBufferFull; 4532} 4533 4534status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4535{ 4536 int active; 4537 status_t result; 4538 audio_track_cblk_t* cblk = mCblk; 4539 uint32_t framesReq = buffer->frameCount; 4540 4541// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4542 buffer->frameCount = 0; 4543 4544 uint32_t framesAvail = cblk->framesAvailable(); 4545 4546 4547 if (framesAvail == 0) { 4548 Mutex::Autolock _l(cblk->lock); 4549 goto start_loop_here; 4550 while (framesAvail == 0) { 4551 active = mActive; 4552 if (CC_UNLIKELY(!active)) { 4553 ALOGV("Not active and NO_MORE_BUFFERS"); 4554 return NO_MORE_BUFFERS; 4555 } 4556 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4557 if (result != NO_ERROR) { 4558 return NO_MORE_BUFFERS; 4559 } 4560 // read the server count again 4561 start_loop_here: 4562 framesAvail = cblk->framesAvailable_l(); 4563 } 4564 } 4565 4566// if (framesAvail < framesReq) { 4567// return NO_MORE_BUFFERS; 4568// } 4569 4570 if (framesReq > framesAvail) { 4571 framesReq = framesAvail; 4572 } 4573 4574 uint32_t u = cblk->user; 4575 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4576 4577 if (framesReq > bufferEnd - u) { 4578 framesReq = bufferEnd - u; 4579 } 4580 4581 buffer->frameCount = framesReq; 4582 buffer->raw = (void *)cblk->buffer(u); 4583 return NO_ERROR; 4584} 4585 4586 4587void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4588{ 4589 size_t size = mBufferQueue.size(); 4590 4591 for (size_t i = 0; i < size; i++) { 4592 Buffer *pBuffer = mBufferQueue.itemAt(i); 4593 delete [] pBuffer->mBuffer; 4594 delete pBuffer; 4595 } 4596 mBufferQueue.clear(); 4597} 4598 4599// ---------------------------------------------------------------------------- 4600 4601AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4602 : RefBase(), 4603 mAudioFlinger(audioFlinger), 4604 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4605 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4606 mPid(pid), 4607 mTimedTrackCount(0) 4608{ 4609 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4610} 4611 4612// Client destructor must be called with AudioFlinger::mLock held 4613AudioFlinger::Client::~Client() 4614{ 4615 mAudioFlinger->removeClient_l(mPid); 4616} 4617 4618sp<MemoryDealer> AudioFlinger::Client::heap() const 4619{ 4620 return mMemoryDealer; 4621} 4622 4623// Reserve one of the limited slots for a timed audio track associated 4624// with this client 4625bool AudioFlinger::Client::reserveTimedTrack() 4626{ 4627 const int kMaxTimedTracksPerClient = 4; 4628 4629 Mutex::Autolock _l(mTimedTrackLock); 4630 4631 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4632 ALOGW("can not create timed track - pid %d has exceeded the limit", 4633 mPid); 4634 return false; 4635 } 4636 4637 mTimedTrackCount++; 4638 return true; 4639} 4640 4641// Release a slot for a timed audio track 4642void AudioFlinger::Client::releaseTimedTrack() 4643{ 4644 Mutex::Autolock _l(mTimedTrackLock); 4645 mTimedTrackCount--; 4646} 4647 4648// ---------------------------------------------------------------------------- 4649 4650AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4651 const sp<IAudioFlingerClient>& client, 4652 pid_t pid) 4653 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4654{ 4655} 4656 4657AudioFlinger::NotificationClient::~NotificationClient() 4658{ 4659} 4660 4661void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4662{ 4663 sp<NotificationClient> keep(this); 4664 mAudioFlinger->removeNotificationClient(mPid); 4665} 4666 4667// ---------------------------------------------------------------------------- 4668 4669AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4670 : BnAudioTrack(), 4671 mTrack(track) 4672{ 4673} 4674 4675AudioFlinger::TrackHandle::~TrackHandle() { 4676 // just stop the track on deletion, associated resources 4677 // will be freed from the main thread once all pending buffers have 4678 // been played. Unless it's not in the active track list, in which 4679 // case we free everything now... 4680 mTrack->destroy(); 4681} 4682 4683sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4684 return mTrack->getCblk(); 4685} 4686 4687status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4688 return mTrack->start(tid); 4689} 4690 4691void AudioFlinger::TrackHandle::stop() { 4692 mTrack->stop(); 4693} 4694 4695void AudioFlinger::TrackHandle::flush() { 4696 mTrack->flush(); 4697} 4698 4699void AudioFlinger::TrackHandle::mute(bool e) { 4700 mTrack->mute(e); 4701} 4702 4703void AudioFlinger::TrackHandle::pause() { 4704 mTrack->pause(); 4705} 4706 4707status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4708{ 4709 return mTrack->attachAuxEffect(EffectId); 4710} 4711 4712status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4713 sp<IMemory>* buffer) { 4714 if (!mTrack->isTimedTrack()) 4715 return INVALID_OPERATION; 4716 4717 PlaybackThread::TimedTrack* tt = 4718 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4719 return tt->allocateTimedBuffer(size, buffer); 4720} 4721 4722status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4723 int64_t pts) { 4724 if (!mTrack->isTimedTrack()) 4725 return INVALID_OPERATION; 4726 4727 PlaybackThread::TimedTrack* tt = 4728 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4729 return tt->queueTimedBuffer(buffer, pts); 4730} 4731 4732status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4733 const LinearTransform& xform, int target) { 4734 4735 if (!mTrack->isTimedTrack()) 4736 return INVALID_OPERATION; 4737 4738 PlaybackThread::TimedTrack* tt = 4739 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4740 return tt->setMediaTimeTransform( 4741 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4742} 4743 4744status_t AudioFlinger::TrackHandle::onTransact( 4745 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4746{ 4747 return BnAudioTrack::onTransact(code, data, reply, flags); 4748} 4749 4750// ---------------------------------------------------------------------------- 4751 4752sp<IAudioRecord> AudioFlinger::openRecord( 4753 pid_t pid, 4754 audio_io_handle_t input, 4755 uint32_t sampleRate, 4756 audio_format_t format, 4757 uint32_t channelMask, 4758 int frameCount, 4759 IAudioFlinger::track_flags_t flags, 4760 int *sessionId, 4761 status_t *status) 4762{ 4763 sp<RecordThread::RecordTrack> recordTrack; 4764 sp<RecordHandle> recordHandle; 4765 sp<Client> client; 4766 status_t lStatus; 4767 RecordThread *thread; 4768 size_t inFrameCount; 4769 int lSessionId; 4770 4771 // check calling permissions 4772 if (!recordingAllowed()) { 4773 lStatus = PERMISSION_DENIED; 4774 goto Exit; 4775 } 4776 4777 // add client to list 4778 { // scope for mLock 4779 Mutex::Autolock _l(mLock); 4780 thread = checkRecordThread_l(input); 4781 if (thread == NULL) { 4782 lStatus = BAD_VALUE; 4783 goto Exit; 4784 } 4785 4786 client = registerPid_l(pid); 4787 4788 // If no audio session id is provided, create one here 4789 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4790 lSessionId = *sessionId; 4791 } else { 4792 lSessionId = nextUniqueId(); 4793 if (sessionId != NULL) { 4794 *sessionId = lSessionId; 4795 } 4796 } 4797 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4798 recordTrack = thread->createRecordTrack_l(client, 4799 sampleRate, 4800 format, 4801 channelMask, 4802 frameCount, 4803 lSessionId, 4804 &lStatus); 4805 } 4806 if (lStatus != NO_ERROR) { 4807 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4808 // destructor is called by the TrackBase destructor with mLock held 4809 client.clear(); 4810 recordTrack.clear(); 4811 goto Exit; 4812 } 4813 4814 // return to handle to client 4815 recordHandle = new RecordHandle(recordTrack); 4816 lStatus = NO_ERROR; 4817 4818Exit: 4819 if (status) { 4820 *status = lStatus; 4821 } 4822 return recordHandle; 4823} 4824 4825// ---------------------------------------------------------------------------- 4826 4827AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4828 : BnAudioRecord(), 4829 mRecordTrack(recordTrack) 4830{ 4831} 4832 4833AudioFlinger::RecordHandle::~RecordHandle() { 4834 stop(); 4835} 4836 4837sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4838 return mRecordTrack->getCblk(); 4839} 4840 4841status_t AudioFlinger::RecordHandle::start(pid_t tid, int event, int triggerSession) { 4842 ALOGV("RecordHandle::start()"); 4843 return mRecordTrack->start(tid, (AudioSystem::sync_event_t)event, triggerSession); 4844} 4845 4846void AudioFlinger::RecordHandle::stop() { 4847 ALOGV("RecordHandle::stop()"); 4848 mRecordTrack->stop(); 4849} 4850 4851status_t AudioFlinger::RecordHandle::onTransact( 4852 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4853{ 4854 return BnAudioRecord::onTransact(code, data, reply, flags); 4855} 4856 4857// ---------------------------------------------------------------------------- 4858 4859AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4860 AudioStreamIn *input, 4861 uint32_t sampleRate, 4862 uint32_t channels, 4863 audio_io_handle_t id, 4864 uint32_t device) : 4865 ThreadBase(audioFlinger, id, device, RECORD), 4866 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4867 // mRsmpInIndex and mInputBytes set by readInputParameters() 4868 mReqChannelCount(popcount(channels)), 4869 mReqSampleRate(sampleRate) 4870 // mBytesRead is only meaningful while active, and so is cleared in start() 4871 // (but might be better to also clear here for dump?) 4872{ 4873 snprintf(mName, kNameLength, "AudioIn_%X", id); 4874 4875 readInputParameters(); 4876} 4877 4878 4879AudioFlinger::RecordThread::~RecordThread() 4880{ 4881 delete[] mRsmpInBuffer; 4882 delete mResampler; 4883 delete[] mRsmpOutBuffer; 4884} 4885 4886void AudioFlinger::RecordThread::onFirstRef() 4887{ 4888 run(mName, PRIORITY_URGENT_AUDIO); 4889} 4890 4891status_t AudioFlinger::RecordThread::readyToRun() 4892{ 4893 status_t status = initCheck(); 4894 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4895 return status; 4896} 4897 4898bool AudioFlinger::RecordThread::threadLoop() 4899{ 4900 AudioBufferProvider::Buffer buffer; 4901 sp<RecordTrack> activeTrack; 4902 Vector< sp<EffectChain> > effectChains; 4903 4904 nsecs_t lastWarning = 0; 4905 4906 acquireWakeLock(); 4907 4908 // start recording 4909 while (!exitPending()) { 4910 4911 processConfigEvents(); 4912 4913 { // scope for mLock 4914 Mutex::Autolock _l(mLock); 4915 checkForNewParameters_l(); 4916 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4917 if (!mStandby) { 4918 mInput->stream->common.standby(&mInput->stream->common); 4919 mStandby = true; 4920 } 4921 4922 if (exitPending()) break; 4923 4924 releaseWakeLock_l(); 4925 ALOGV("RecordThread: loop stopping"); 4926 // go to sleep 4927 mWaitWorkCV.wait(mLock); 4928 ALOGV("RecordThread: loop starting"); 4929 acquireWakeLock_l(); 4930 continue; 4931 } 4932 if (mActiveTrack != 0) { 4933 if (mActiveTrack->mState == TrackBase::PAUSING) { 4934 if (!mStandby) { 4935 mInput->stream->common.standby(&mInput->stream->common); 4936 mStandby = true; 4937 } 4938 mActiveTrack.clear(); 4939 mStartStopCond.broadcast(); 4940 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4941 if (mReqChannelCount != mActiveTrack->channelCount()) { 4942 mActiveTrack.clear(); 4943 mStartStopCond.broadcast(); 4944 } else if (mBytesRead != 0) { 4945 // record start succeeds only if first read from audio input 4946 // succeeds 4947 if (mBytesRead > 0) { 4948 mActiveTrack->mState = TrackBase::ACTIVE; 4949 } else { 4950 mActiveTrack.clear(); 4951 } 4952 mStartStopCond.broadcast(); 4953 } 4954 mStandby = false; 4955 } 4956 } 4957 lockEffectChains_l(effectChains); 4958 } 4959 4960 if (mActiveTrack != 0) { 4961 if (mActiveTrack->mState != TrackBase::ACTIVE && 4962 mActiveTrack->mState != TrackBase::RESUMING) { 4963 unlockEffectChains(effectChains); 4964 usleep(kRecordThreadSleepUs); 4965 continue; 4966 } 4967 for (size_t i = 0; i < effectChains.size(); i ++) { 4968 effectChains[i]->process_l(); 4969 } 4970 4971 buffer.frameCount = mFrameCount; 4972 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4973 size_t framesOut = buffer.frameCount; 4974 if (mResampler == NULL) { 4975 // no resampling 4976 while (framesOut) { 4977 size_t framesIn = mFrameCount - mRsmpInIndex; 4978 if (framesIn) { 4979 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4980 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4981 if (framesIn > framesOut) 4982 framesIn = framesOut; 4983 mRsmpInIndex += framesIn; 4984 framesOut -= framesIn; 4985 if ((int)mChannelCount == mReqChannelCount || 4986 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4987 memcpy(dst, src, framesIn * mFrameSize); 4988 } else { 4989 int16_t *src16 = (int16_t *)src; 4990 int16_t *dst16 = (int16_t *)dst; 4991 if (mChannelCount == 1) { 4992 while (framesIn--) { 4993 *dst16++ = *src16; 4994 *dst16++ = *src16++; 4995 } 4996 } else { 4997 while (framesIn--) { 4998 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4999 src16 += 2; 5000 } 5001 } 5002 } 5003 } 5004 if (framesOut && mFrameCount == mRsmpInIndex) { 5005 if (framesOut == mFrameCount && 5006 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5007 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5008 framesOut = 0; 5009 } else { 5010 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5011 mRsmpInIndex = 0; 5012 } 5013 if (mBytesRead < 0) { 5014 ALOGE("Error reading audio input"); 5015 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5016 // Force input into standby so that it tries to 5017 // recover at next read attempt 5018 mInput->stream->common.standby(&mInput->stream->common); 5019 usleep(kRecordThreadSleepUs); 5020 } 5021 mRsmpInIndex = mFrameCount; 5022 framesOut = 0; 5023 buffer.frameCount = 0; 5024 } 5025 } 5026 } 5027 } else { 5028 // resampling 5029 5030 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5031 // alter output frame count as if we were expecting stereo samples 5032 if (mChannelCount == 1 && mReqChannelCount == 1) { 5033 framesOut >>= 1; 5034 } 5035 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5036 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5037 // are 32 bit aligned which should be always true. 5038 if (mChannelCount == 2 && mReqChannelCount == 1) { 5039 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5040 // the resampler always outputs stereo samples: do post stereo to mono conversion 5041 int16_t *src = (int16_t *)mRsmpOutBuffer; 5042 int16_t *dst = buffer.i16; 5043 while (framesOut--) { 5044 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5045 src += 2; 5046 } 5047 } else { 5048 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5049 } 5050 5051 } 5052 if (mFramestoDrop == 0) { 5053 mActiveTrack->releaseBuffer(&buffer); 5054 } else { 5055 if (mFramestoDrop > 0) { 5056 mFramestoDrop -= buffer.frameCount; 5057 if (mFramestoDrop < 0) { 5058 mFramestoDrop = 0; 5059 } 5060 } 5061 } 5062 mActiveTrack->overflow(); 5063 } 5064 // client isn't retrieving buffers fast enough 5065 else { 5066 if (!mActiveTrack->setOverflow()) { 5067 nsecs_t now = systemTime(); 5068 if ((now - lastWarning) > kWarningThrottleNs) { 5069 ALOGW("RecordThread: buffer overflow"); 5070 lastWarning = now; 5071 } 5072 } 5073 // Release the processor for a while before asking for a new buffer. 5074 // This will give the application more chance to read from the buffer and 5075 // clear the overflow. 5076 usleep(kRecordThreadSleepUs); 5077 } 5078 } 5079 // enable changes in effect chain 5080 unlockEffectChains(effectChains); 5081 effectChains.clear(); 5082 } 5083 5084 if (!mStandby) { 5085 mInput->stream->common.standby(&mInput->stream->common); 5086 } 5087 mActiveTrack.clear(); 5088 5089 mStartStopCond.broadcast(); 5090 5091 releaseWakeLock(); 5092 5093 ALOGV("RecordThread %p exiting", this); 5094 return false; 5095} 5096 5097 5098sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5099 const sp<AudioFlinger::Client>& client, 5100 uint32_t sampleRate, 5101 audio_format_t format, 5102 int channelMask, 5103 int frameCount, 5104 int sessionId, 5105 status_t *status) 5106{ 5107 sp<RecordTrack> track; 5108 status_t lStatus; 5109 5110 lStatus = initCheck(); 5111 if (lStatus != NO_ERROR) { 5112 ALOGE("Audio driver not initialized."); 5113 goto Exit; 5114 } 5115 5116 { // scope for mLock 5117 Mutex::Autolock _l(mLock); 5118 5119 track = new RecordTrack(this, client, sampleRate, 5120 format, channelMask, frameCount, sessionId); 5121 5122 if (track->getCblk() == 0) { 5123 lStatus = NO_MEMORY; 5124 goto Exit; 5125 } 5126 5127 mTrack = track.get(); 5128 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5129 bool suspend = audio_is_bluetooth_sco_device( 5130 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5131 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5132 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5133 } 5134 lStatus = NO_ERROR; 5135 5136Exit: 5137 if (status) { 5138 *status = lStatus; 5139 } 5140 return track; 5141} 5142 5143status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5144 pid_t tid, AudioSystem::sync_event_t event, 5145 int triggerSession) 5146{ 5147 ALOGV("RecordThread::start tid=%d, event %d, triggerSession %d", tid, event, triggerSession); 5148 sp<ThreadBase> strongMe = this; 5149 status_t status = NO_ERROR; 5150 5151 if (event == AudioSystem::SYNC_EVENT_NONE) { 5152 mSyncStartEvent.clear(); 5153 mFramestoDrop = 0; 5154 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5155 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5156 triggerSession, 5157 recordTrack->sessionId(), 5158 syncStartEventCallback, 5159 this); 5160 mFramestoDrop = -1; 5161 } 5162 5163 { 5164 AutoMutex lock(mLock); 5165 if (mActiveTrack != 0) { 5166 if (recordTrack != mActiveTrack.get()) { 5167 status = -EBUSY; 5168 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5169 mActiveTrack->mState = TrackBase::ACTIVE; 5170 } 5171 return status; 5172 } 5173 5174 recordTrack->mState = TrackBase::IDLE; 5175 mActiveTrack = recordTrack; 5176 mLock.unlock(); 5177 status_t status = AudioSystem::startInput(mId); 5178 mLock.lock(); 5179 if (status != NO_ERROR) { 5180 mActiveTrack.clear(); 5181 clearSyncStartEvent(); 5182 return status; 5183 } 5184 mRsmpInIndex = mFrameCount; 5185 mBytesRead = 0; 5186 if (mResampler != NULL) { 5187 mResampler->reset(); 5188 } 5189 mActiveTrack->mState = TrackBase::RESUMING; 5190 // signal thread to start 5191 ALOGV("Signal record thread"); 5192 mWaitWorkCV.signal(); 5193 // do not wait for mStartStopCond if exiting 5194 if (exitPending()) { 5195 mActiveTrack.clear(); 5196 status = INVALID_OPERATION; 5197 goto startError; 5198 } 5199 mStartStopCond.wait(mLock); 5200 if (mActiveTrack == 0) { 5201 ALOGV("Record failed to start"); 5202 status = BAD_VALUE; 5203 goto startError; 5204 } 5205 ALOGV("Record started OK"); 5206 return status; 5207 } 5208startError: 5209 AudioSystem::stopInput(mId); 5210 clearSyncStartEvent(); 5211 return status; 5212} 5213 5214void AudioFlinger::RecordThread::clearSyncStartEvent() 5215{ 5216 if (mSyncStartEvent != 0) { 5217 mSyncStartEvent->cancel(); 5218 } 5219 mSyncStartEvent.clear(); 5220} 5221 5222void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5223{ 5224 sp<SyncEvent> strongEvent = event.promote(); 5225 5226 if (strongEvent != 0) { 5227 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5228 me->handleSyncStartEvent(strongEvent); 5229 } 5230} 5231 5232void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5233{ 5234 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5235 mActiveTrack.get(), 5236 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5237 event->listenerSession()); 5238 5239 if (mActiveTrack != 0 && 5240 event == mSyncStartEvent) { 5241 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5242 // from audio HAL 5243 mFramestoDrop = mFrameCount * 2; 5244 mSyncStartEvent.clear(); 5245 } 5246} 5247 5248void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5249 ALOGV("RecordThread::stop"); 5250 sp<ThreadBase> strongMe = this; 5251 { 5252 AutoMutex lock(mLock); 5253 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5254 mActiveTrack->mState = TrackBase::PAUSING; 5255 // do not wait for mStartStopCond if exiting 5256 if (exitPending()) { 5257 return; 5258 } 5259 mStartStopCond.wait(mLock); 5260 // if we have been restarted, recordTrack == mActiveTrack.get() here 5261 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5262 mLock.unlock(); 5263 AudioSystem::stopInput(mId); 5264 mLock.lock(); 5265 ALOGV("Record stopped OK"); 5266 } 5267 } 5268 } 5269} 5270 5271bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5272{ 5273 return false; 5274} 5275 5276status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5277{ 5278 if (!isValidSyncEvent(event)) { 5279 return BAD_VALUE; 5280 } 5281 5282 Mutex::Autolock _l(mLock); 5283 5284 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5285 mTrack->setSyncEvent(event); 5286 return NO_ERROR; 5287 } 5288 return NAME_NOT_FOUND; 5289} 5290 5291status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5292{ 5293 const size_t SIZE = 256; 5294 char buffer[SIZE]; 5295 String8 result; 5296 5297 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5298 result.append(buffer); 5299 5300 if (mActiveTrack != 0) { 5301 result.append("Active Track:\n"); 5302 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5303 mActiveTrack->dump(buffer, SIZE); 5304 result.append(buffer); 5305 5306 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5307 result.append(buffer); 5308 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5309 result.append(buffer); 5310 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5311 result.append(buffer); 5312 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5313 result.append(buffer); 5314 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5315 result.append(buffer); 5316 5317 5318 } else { 5319 result.append("No record client\n"); 5320 } 5321 write(fd, result.string(), result.size()); 5322 5323 dumpBase(fd, args); 5324 dumpEffectChains(fd, args); 5325 5326 return NO_ERROR; 5327} 5328 5329// AudioBufferProvider interface 5330status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5331{ 5332 size_t framesReq = buffer->frameCount; 5333 size_t framesReady = mFrameCount - mRsmpInIndex; 5334 int channelCount; 5335 5336 if (framesReady == 0) { 5337 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5338 if (mBytesRead < 0) { 5339 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5340 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5341 // Force input into standby so that it tries to 5342 // recover at next read attempt 5343 mInput->stream->common.standby(&mInput->stream->common); 5344 usleep(kRecordThreadSleepUs); 5345 } 5346 buffer->raw = NULL; 5347 buffer->frameCount = 0; 5348 return NOT_ENOUGH_DATA; 5349 } 5350 mRsmpInIndex = 0; 5351 framesReady = mFrameCount; 5352 } 5353 5354 if (framesReq > framesReady) { 5355 framesReq = framesReady; 5356 } 5357 5358 if (mChannelCount == 1 && mReqChannelCount == 2) { 5359 channelCount = 1; 5360 } else { 5361 channelCount = 2; 5362 } 5363 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5364 buffer->frameCount = framesReq; 5365 return NO_ERROR; 5366} 5367 5368// AudioBufferProvider interface 5369void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5370{ 5371 mRsmpInIndex += buffer->frameCount; 5372 buffer->frameCount = 0; 5373} 5374 5375bool AudioFlinger::RecordThread::checkForNewParameters_l() 5376{ 5377 bool reconfig = false; 5378 5379 while (!mNewParameters.isEmpty()) { 5380 status_t status = NO_ERROR; 5381 String8 keyValuePair = mNewParameters[0]; 5382 AudioParameter param = AudioParameter(keyValuePair); 5383 int value; 5384 audio_format_t reqFormat = mFormat; 5385 int reqSamplingRate = mReqSampleRate; 5386 int reqChannelCount = mReqChannelCount; 5387 5388 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5389 reqSamplingRate = value; 5390 reconfig = true; 5391 } 5392 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5393 reqFormat = (audio_format_t) value; 5394 reconfig = true; 5395 } 5396 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5397 reqChannelCount = popcount(value); 5398 reconfig = true; 5399 } 5400 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5401 // do not accept frame count changes if tracks are open as the track buffer 5402 // size depends on frame count and correct behavior would not be guaranteed 5403 // if frame count is changed after track creation 5404 if (mActiveTrack != 0) { 5405 status = INVALID_OPERATION; 5406 } else { 5407 reconfig = true; 5408 } 5409 } 5410 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5411 // forward device change to effects that have requested to be 5412 // aware of attached audio device. 5413 for (size_t i = 0; i < mEffectChains.size(); i++) { 5414 mEffectChains[i]->setDevice_l(value); 5415 } 5416 // store input device and output device but do not forward output device to audio HAL. 5417 // Note that status is ignored by the caller for output device 5418 // (see AudioFlinger::setParameters() 5419 if (value & AUDIO_DEVICE_OUT_ALL) { 5420 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5421 status = BAD_VALUE; 5422 } else { 5423 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5424 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5425 if (mTrack != NULL) { 5426 bool suspend = audio_is_bluetooth_sco_device( 5427 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5428 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5429 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5430 } 5431 } 5432 mDevice |= (uint32_t)value; 5433 } 5434 if (status == NO_ERROR) { 5435 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5436 if (status == INVALID_OPERATION) { 5437 mInput->stream->common.standby(&mInput->stream->common); 5438 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5439 keyValuePair.string()); 5440 } 5441 if (reconfig) { 5442 if (status == BAD_VALUE && 5443 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5444 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5445 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5446 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5447 (reqChannelCount <= FCC_2)) { 5448 status = NO_ERROR; 5449 } 5450 if (status == NO_ERROR) { 5451 readInputParameters(); 5452 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5453 } 5454 } 5455 } 5456 5457 mNewParameters.removeAt(0); 5458 5459 mParamStatus = status; 5460 mParamCond.signal(); 5461 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5462 // already timed out waiting for the status and will never signal the condition. 5463 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5464 } 5465 return reconfig; 5466} 5467 5468String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5469{ 5470 char *s; 5471 String8 out_s8 = String8(); 5472 5473 Mutex::Autolock _l(mLock); 5474 if (initCheck() != NO_ERROR) { 5475 return out_s8; 5476 } 5477 5478 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5479 out_s8 = String8(s); 5480 free(s); 5481 return out_s8; 5482} 5483 5484void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5485 AudioSystem::OutputDescriptor desc; 5486 void *param2 = NULL; 5487 5488 switch (event) { 5489 case AudioSystem::INPUT_OPENED: 5490 case AudioSystem::INPUT_CONFIG_CHANGED: 5491 desc.channels = mChannelMask; 5492 desc.samplingRate = mSampleRate; 5493 desc.format = mFormat; 5494 desc.frameCount = mFrameCount; 5495 desc.latency = 0; 5496 param2 = &desc; 5497 break; 5498 5499 case AudioSystem::INPUT_CLOSED: 5500 default: 5501 break; 5502 } 5503 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5504} 5505 5506void AudioFlinger::RecordThread::readInputParameters() 5507{ 5508 delete mRsmpInBuffer; 5509 // mRsmpInBuffer is always assigned a new[] below 5510 delete mRsmpOutBuffer; 5511 mRsmpOutBuffer = NULL; 5512 delete mResampler; 5513 mResampler = NULL; 5514 5515 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5516 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5517 mChannelCount = (uint16_t)popcount(mChannelMask); 5518 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5519 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5520 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5521 mFrameCount = mInputBytes / mFrameSize; 5522 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5523 5524 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5525 { 5526 int channelCount; 5527 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5528 // stereo to mono post process as the resampler always outputs stereo. 5529 if (mChannelCount == 1 && mReqChannelCount == 2) { 5530 channelCount = 1; 5531 } else { 5532 channelCount = 2; 5533 } 5534 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5535 mResampler->setSampleRate(mSampleRate); 5536 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5537 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5538 5539 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5540 if (mChannelCount == 1 && mReqChannelCount == 1) { 5541 mFrameCount >>= 1; 5542 } 5543 5544 } 5545 mRsmpInIndex = mFrameCount; 5546} 5547 5548unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5549{ 5550 Mutex::Autolock _l(mLock); 5551 if (initCheck() != NO_ERROR) { 5552 return 0; 5553 } 5554 5555 return mInput->stream->get_input_frames_lost(mInput->stream); 5556} 5557 5558uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5559{ 5560 Mutex::Autolock _l(mLock); 5561 uint32_t result = 0; 5562 if (getEffectChain_l(sessionId) != 0) { 5563 result = EFFECT_SESSION; 5564 } 5565 5566 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5567 result |= TRACK_SESSION; 5568 } 5569 5570 return result; 5571} 5572 5573AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5574{ 5575 Mutex::Autolock _l(mLock); 5576 return mTrack; 5577} 5578 5579AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5580{ 5581 Mutex::Autolock _l(mLock); 5582 return mInput; 5583} 5584 5585AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5586{ 5587 Mutex::Autolock _l(mLock); 5588 AudioStreamIn *input = mInput; 5589 mInput = NULL; 5590 return input; 5591} 5592 5593// this method must always be called either with ThreadBase mLock held or inside the thread loop 5594audio_stream_t* AudioFlinger::RecordThread::stream() const 5595{ 5596 if (mInput == NULL) { 5597 return NULL; 5598 } 5599 return &mInput->stream->common; 5600} 5601 5602 5603// ---------------------------------------------------------------------------- 5604 5605audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 5606{ 5607 if (!settingsAllowed()) { 5608 return 0; 5609 } 5610 Mutex::Autolock _l(mLock); 5611 return loadHwModule_l(name); 5612} 5613 5614// loadHwModule_l() must be called with AudioFlinger::mLock held 5615audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 5616{ 5617 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5618 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 5619 ALOGW("loadHwModule() module %s already loaded", name); 5620 return mAudioHwDevs.keyAt(i); 5621 } 5622 } 5623 5624 const hw_module_t *mod; 5625 audio_hw_device_t *dev; 5626 5627 int rc = load_audio_interface(name, &mod, &dev); 5628 if (rc) { 5629 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 5630 return 0; 5631 } 5632 5633 mHardwareStatus = AUDIO_HW_INIT; 5634 rc = dev->init_check(dev); 5635 mHardwareStatus = AUDIO_HW_IDLE; 5636 if (rc) { 5637 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 5638 return 0; 5639 } 5640 5641 if ((mMasterVolumeSupportLvl != MVS_NONE) && 5642 (NULL != dev->set_master_volume)) { 5643 AutoMutex lock(mHardwareLock); 5644 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5645 dev->set_master_volume(dev, mMasterVolume); 5646 mHardwareStatus = AUDIO_HW_IDLE; 5647 } 5648 5649 audio_module_handle_t handle = nextUniqueId(); 5650 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 5651 5652 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 5653 name, mod->name, mod->id, handle); 5654 5655 return handle; 5656 5657} 5658 5659audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 5660 audio_devices_t *pDevices, 5661 uint32_t *pSamplingRate, 5662 audio_format_t *pFormat, 5663 audio_channel_mask_t *pChannelMask, 5664 uint32_t *pLatencyMs, 5665 audio_policy_output_flags_t flags) 5666{ 5667 status_t status; 5668 PlaybackThread *thread = NULL; 5669 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5670 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5671 audio_channel_mask_t channelMask = pChannelMask ? *pChannelMask : 0; 5672 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5673 audio_stream_out_t *outStream; 5674 audio_hw_device_t *outHwDev; 5675 5676 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5677 module, 5678 pDevices ? *pDevices : 0, 5679 samplingRate, 5680 format, 5681 channelMask, 5682 flags); 5683 5684 if (pDevices == NULL || *pDevices == 0) { 5685 return 0; 5686 } 5687 5688 Mutex::Autolock _l(mLock); 5689 5690 outHwDev = findSuitableHwDev_l(module, *pDevices); 5691 if (outHwDev == NULL) 5692 return 0; 5693 5694 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5695 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5696 &channelMask, &samplingRate, &outStream); 5697 mHardwareStatus = AUDIO_HW_IDLE; 5698 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5699 outStream, 5700 samplingRate, 5701 format, 5702 channelMask, 5703 status); 5704 5705 if (outStream != NULL) { 5706 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5707 audio_io_handle_t id = nextUniqueId(); 5708 5709 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5710 (format != AUDIO_FORMAT_PCM_16_BIT) || 5711 (channelMask != AUDIO_CHANNEL_OUT_STEREO)) { 5712 thread = new DirectOutputThread(this, output, id, *pDevices); 5713 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5714 } else { 5715 thread = new MixerThread(this, output, id, *pDevices); 5716 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5717 } 5718 mPlaybackThreads.add(id, thread); 5719 5720 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5721 if (pFormat != NULL) *pFormat = format; 5722 if (pChannelMask != NULL) *pChannelMask = channelMask; 5723 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5724 5725 // notify client processes of the new output creation 5726 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5727 5728 // the first primary output opened designates the primary hw device 5729 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_POLICY_OUTPUT_FLAG_PRIMARY)) { 5730 ALOGI("Using module %d has the primary audio interface", module); 5731 mPrimaryHardwareDev = outHwDev; 5732 5733 AutoMutex lock(mHardwareLock); 5734 mHardwareStatus = AUDIO_HW_SET_MODE; 5735 outHwDev->set_mode(outHwDev, mMode); 5736 5737 // Determine the level of master volume support the primary audio HAL has, 5738 // and set the initial master volume at the same time. 5739 float initialVolume = 1.0; 5740 mMasterVolumeSupportLvl = MVS_NONE; 5741 5742 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 5743 if ((NULL != outHwDev->get_master_volume) && 5744 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 5745 mMasterVolumeSupportLvl = MVS_FULL; 5746 } else { 5747 mMasterVolumeSupportLvl = MVS_SETONLY; 5748 initialVolume = 1.0; 5749 } 5750 5751 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5752 if ((NULL == outHwDev->set_master_volume) || 5753 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 5754 mMasterVolumeSupportLvl = MVS_NONE; 5755 } 5756 // now that we have a primary device, initialize master volume on other devices 5757 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5758 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 5759 5760 if ((dev != mPrimaryHardwareDev) && 5761 (NULL != dev->set_master_volume)) { 5762 dev->set_master_volume(dev, initialVolume); 5763 } 5764 } 5765 mHardwareStatus = AUDIO_HW_IDLE; 5766 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 5767 ? initialVolume 5768 : 1.0; 5769 mMasterVolume = initialVolume; 5770 } 5771 return id; 5772 } 5773 5774 return 0; 5775} 5776 5777audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5778 audio_io_handle_t output2) 5779{ 5780 Mutex::Autolock _l(mLock); 5781 MixerThread *thread1 = checkMixerThread_l(output1); 5782 MixerThread *thread2 = checkMixerThread_l(output2); 5783 5784 if (thread1 == NULL || thread2 == NULL) { 5785 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5786 return 0; 5787 } 5788 5789 audio_io_handle_t id = nextUniqueId(); 5790 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5791 thread->addOutputTrack(thread2); 5792 mPlaybackThreads.add(id, thread); 5793 // notify client processes of the new output creation 5794 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5795 return id; 5796} 5797 5798status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5799{ 5800 // keep strong reference on the playback thread so that 5801 // it is not destroyed while exit() is executed 5802 sp<PlaybackThread> thread; 5803 { 5804 Mutex::Autolock _l(mLock); 5805 thread = checkPlaybackThread_l(output); 5806 if (thread == NULL) { 5807 return BAD_VALUE; 5808 } 5809 5810 ALOGV("closeOutput() %d", output); 5811 5812 if (thread->type() == ThreadBase::MIXER) { 5813 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5814 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5815 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5816 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5817 } 5818 } 5819 } 5820 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5821 mPlaybackThreads.removeItem(output); 5822 } 5823 thread->exit(); 5824 // The thread entity (active unit of execution) is no longer running here, 5825 // but the ThreadBase container still exists. 5826 5827 if (thread->type() != ThreadBase::DUPLICATING) { 5828 AudioStreamOut *out = thread->clearOutput(); 5829 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5830 // from now on thread->mOutput is NULL 5831 out->hwDev->close_output_stream(out->hwDev, out->stream); 5832 delete out; 5833 } 5834 return NO_ERROR; 5835} 5836 5837status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5838{ 5839 Mutex::Autolock _l(mLock); 5840 PlaybackThread *thread = checkPlaybackThread_l(output); 5841 5842 if (thread == NULL) { 5843 return BAD_VALUE; 5844 } 5845 5846 ALOGV("suspendOutput() %d", output); 5847 thread->suspend(); 5848 5849 return NO_ERROR; 5850} 5851 5852status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5853{ 5854 Mutex::Autolock _l(mLock); 5855 PlaybackThread *thread = checkPlaybackThread_l(output); 5856 5857 if (thread == NULL) { 5858 return BAD_VALUE; 5859 } 5860 5861 ALOGV("restoreOutput() %d", output); 5862 5863 thread->restore(); 5864 5865 return NO_ERROR; 5866} 5867 5868audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 5869 audio_devices_t *pDevices, 5870 uint32_t *pSamplingRate, 5871 audio_format_t *pFormat, 5872 uint32_t *pChannelMask) 5873{ 5874 status_t status; 5875 RecordThread *thread = NULL; 5876 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5877 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5878 audio_channel_mask_t channelMask = pChannelMask ? *pChannelMask : 0; 5879 uint32_t reqSamplingRate = samplingRate; 5880 audio_format_t reqFormat = format; 5881 audio_channel_mask_t reqChannels = channelMask; 5882 audio_stream_in_t *inStream; 5883 audio_hw_device_t *inHwDev; 5884 5885 if (pDevices == NULL || *pDevices == 0) { 5886 return 0; 5887 } 5888 5889 Mutex::Autolock _l(mLock); 5890 5891 inHwDev = findSuitableHwDev_l(module, *pDevices); 5892 if (inHwDev == NULL) 5893 return 0; 5894 5895 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5896 &channelMask, &samplingRate, 5897 (audio_in_acoustics_t)0, 5898 &inStream); 5899 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 5900 inStream, 5901 samplingRate, 5902 format, 5903 channelMask, 5904 status); 5905 5906 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5907 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5908 // or stereo to mono conversions on 16 bit PCM inputs. 5909 if (inStream == NULL && status == BAD_VALUE && 5910 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5911 (samplingRate <= 2 * reqSamplingRate) && 5912 (popcount(channelMask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5913 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5914 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5915 &channelMask, &samplingRate, 5916 (audio_in_acoustics_t)0, 5917 &inStream); 5918 } 5919 5920 if (inStream != NULL) { 5921 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5922 5923 audio_io_handle_t id = nextUniqueId(); 5924 // Start record thread 5925 // RecorThread require both input and output device indication to forward to audio 5926 // pre processing modules 5927 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5928 thread = new RecordThread(this, 5929 input, 5930 reqSamplingRate, 5931 reqChannels, 5932 id, 5933 device); 5934 mRecordThreads.add(id, thread); 5935 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5936 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5937 if (pFormat != NULL) *pFormat = format; 5938 if (pChannelMask != NULL) *pChannelMask = reqChannels; 5939 5940 input->stream->common.standby(&input->stream->common); 5941 5942 // notify client processes of the new input creation 5943 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5944 return id; 5945 } 5946 5947 return 0; 5948} 5949 5950status_t AudioFlinger::closeInput(audio_io_handle_t input) 5951{ 5952 // keep strong reference on the record thread so that 5953 // it is not destroyed while exit() is executed 5954 sp<RecordThread> thread; 5955 { 5956 Mutex::Autolock _l(mLock); 5957 thread = checkRecordThread_l(input); 5958 if (thread == NULL) { 5959 return BAD_VALUE; 5960 } 5961 5962 ALOGV("closeInput() %d", input); 5963 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5964 mRecordThreads.removeItem(input); 5965 } 5966 thread->exit(); 5967 // The thread entity (active unit of execution) is no longer running here, 5968 // but the ThreadBase container still exists. 5969 5970 AudioStreamIn *in = thread->clearInput(); 5971 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5972 // from now on thread->mInput is NULL 5973 in->hwDev->close_input_stream(in->hwDev, in->stream); 5974 delete in; 5975 5976 return NO_ERROR; 5977} 5978 5979status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5980{ 5981 Mutex::Autolock _l(mLock); 5982 MixerThread *dstThread = checkMixerThread_l(output); 5983 if (dstThread == NULL) { 5984 ALOGW("setStreamOutput() bad output id %d", output); 5985 return BAD_VALUE; 5986 } 5987 5988 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5989 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5990 5991 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5992 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5993 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5994 MixerThread *srcThread = (MixerThread *)thread; 5995 srcThread->invalidateTracks(stream); 5996 } 5997 } 5998 5999 return NO_ERROR; 6000} 6001 6002 6003int AudioFlinger::newAudioSessionId() 6004{ 6005 return nextUniqueId(); 6006} 6007 6008void AudioFlinger::acquireAudioSessionId(int audioSession) 6009{ 6010 Mutex::Autolock _l(mLock); 6011 pid_t caller = IPCThreadState::self()->getCallingPid(); 6012 ALOGV("acquiring %d from %d", audioSession, caller); 6013 size_t num = mAudioSessionRefs.size(); 6014 for (size_t i = 0; i< num; i++) { 6015 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6016 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6017 ref->mCnt++; 6018 ALOGV(" incremented refcount to %d", ref->mCnt); 6019 return; 6020 } 6021 } 6022 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6023 ALOGV(" added new entry for %d", audioSession); 6024} 6025 6026void AudioFlinger::releaseAudioSessionId(int audioSession) 6027{ 6028 Mutex::Autolock _l(mLock); 6029 pid_t caller = IPCThreadState::self()->getCallingPid(); 6030 ALOGV("releasing %d from %d", audioSession, caller); 6031 size_t num = mAudioSessionRefs.size(); 6032 for (size_t i = 0; i< num; i++) { 6033 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6034 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6035 ref->mCnt--; 6036 ALOGV(" decremented refcount to %d", ref->mCnt); 6037 if (ref->mCnt == 0) { 6038 mAudioSessionRefs.removeAt(i); 6039 delete ref; 6040 purgeStaleEffects_l(); 6041 } 6042 return; 6043 } 6044 } 6045 ALOGW("session id %d not found for pid %d", audioSession, caller); 6046} 6047 6048void AudioFlinger::purgeStaleEffects_l() { 6049 6050 ALOGV("purging stale effects"); 6051 6052 Vector< sp<EffectChain> > chains; 6053 6054 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6055 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6056 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6057 sp<EffectChain> ec = t->mEffectChains[j]; 6058 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6059 chains.push(ec); 6060 } 6061 } 6062 } 6063 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6064 sp<RecordThread> t = mRecordThreads.valueAt(i); 6065 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6066 sp<EffectChain> ec = t->mEffectChains[j]; 6067 chains.push(ec); 6068 } 6069 } 6070 6071 for (size_t i = 0; i < chains.size(); i++) { 6072 sp<EffectChain> ec = chains[i]; 6073 int sessionid = ec->sessionId(); 6074 sp<ThreadBase> t = ec->mThread.promote(); 6075 if (t == 0) { 6076 continue; 6077 } 6078 size_t numsessionrefs = mAudioSessionRefs.size(); 6079 bool found = false; 6080 for (size_t k = 0; k < numsessionrefs; k++) { 6081 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6082 if (ref->mSessionid == sessionid) { 6083 ALOGV(" session %d still exists for %d with %d refs", 6084 sessionid, ref->mPid, ref->mCnt); 6085 found = true; 6086 break; 6087 } 6088 } 6089 if (!found) { 6090 // remove all effects from the chain 6091 while (ec->mEffects.size()) { 6092 sp<EffectModule> effect = ec->mEffects[0]; 6093 effect->unPin(); 6094 Mutex::Autolock _l (t->mLock); 6095 t->removeEffect_l(effect); 6096 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6097 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6098 if (handle != 0) { 6099 handle->mEffect.clear(); 6100 if (handle->mHasControl && handle->mEnabled) { 6101 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6102 } 6103 } 6104 } 6105 AudioSystem::unregisterEffect(effect->id()); 6106 } 6107 } 6108 } 6109 return; 6110} 6111 6112// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6113AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6114{ 6115 return mPlaybackThreads.valueFor(output).get(); 6116} 6117 6118// checkMixerThread_l() must be called with AudioFlinger::mLock held 6119AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6120{ 6121 PlaybackThread *thread = checkPlaybackThread_l(output); 6122 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6123} 6124 6125// checkRecordThread_l() must be called with AudioFlinger::mLock held 6126AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6127{ 6128 return mRecordThreads.valueFor(input).get(); 6129} 6130 6131uint32_t AudioFlinger::nextUniqueId() 6132{ 6133 return android_atomic_inc(&mNextUniqueId); 6134} 6135 6136AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6137{ 6138 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6139 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6140 AudioStreamOut *output = thread->getOutput(); 6141 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6142 return thread; 6143 } 6144 } 6145 return NULL; 6146} 6147 6148uint32_t AudioFlinger::primaryOutputDevice_l() const 6149{ 6150 PlaybackThread *thread = primaryPlaybackThread_l(); 6151 6152 if (thread == NULL) { 6153 return 0; 6154 } 6155 6156 return thread->device(); 6157} 6158 6159sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6160 int triggerSession, 6161 int listenerSession, 6162 sync_event_callback_t callBack, 6163 void *cookie) 6164{ 6165 Mutex::Autolock _l(mLock); 6166 6167 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6168 status_t playStatus = NAME_NOT_FOUND; 6169 status_t recStatus = NAME_NOT_FOUND; 6170 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6171 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6172 if (playStatus == NO_ERROR) { 6173 return event; 6174 } 6175 } 6176 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6177 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6178 if (recStatus == NO_ERROR) { 6179 return event; 6180 } 6181 } 6182 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6183 mPendingSyncEvents.add(event); 6184 } else { 6185 ALOGV("createSyncEvent() invalid event %d", event->type()); 6186 event.clear(); 6187 } 6188 return event; 6189} 6190 6191// ---------------------------------------------------------------------------- 6192// Effect management 6193// ---------------------------------------------------------------------------- 6194 6195 6196status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6197{ 6198 Mutex::Autolock _l(mLock); 6199 return EffectQueryNumberEffects(numEffects); 6200} 6201 6202status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6203{ 6204 Mutex::Autolock _l(mLock); 6205 return EffectQueryEffect(index, descriptor); 6206} 6207 6208status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6209 effect_descriptor_t *descriptor) const 6210{ 6211 Mutex::Autolock _l(mLock); 6212 return EffectGetDescriptor(pUuid, descriptor); 6213} 6214 6215 6216sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6217 effect_descriptor_t *pDesc, 6218 const sp<IEffectClient>& effectClient, 6219 int32_t priority, 6220 audio_io_handle_t io, 6221 int sessionId, 6222 status_t *status, 6223 int *id, 6224 int *enabled) 6225{ 6226 status_t lStatus = NO_ERROR; 6227 sp<EffectHandle> handle; 6228 effect_descriptor_t desc; 6229 6230 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6231 pid, effectClient.get(), priority, sessionId, io); 6232 6233 if (pDesc == NULL) { 6234 lStatus = BAD_VALUE; 6235 goto Exit; 6236 } 6237 6238 // check audio settings permission for global effects 6239 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6240 lStatus = PERMISSION_DENIED; 6241 goto Exit; 6242 } 6243 6244 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6245 // that can only be created by audio policy manager (running in same process) 6246 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6247 lStatus = PERMISSION_DENIED; 6248 goto Exit; 6249 } 6250 6251 if (io == 0) { 6252 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6253 // output must be specified by AudioPolicyManager when using session 6254 // AUDIO_SESSION_OUTPUT_STAGE 6255 lStatus = BAD_VALUE; 6256 goto Exit; 6257 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6258 // if the output returned by getOutputForEffect() is removed before we lock the 6259 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6260 // and we will exit safely 6261 io = AudioSystem::getOutputForEffect(&desc); 6262 } 6263 } 6264 6265 { 6266 Mutex::Autolock _l(mLock); 6267 6268 6269 if (!EffectIsNullUuid(&pDesc->uuid)) { 6270 // if uuid is specified, request effect descriptor 6271 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6272 if (lStatus < 0) { 6273 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6274 goto Exit; 6275 } 6276 } else { 6277 // if uuid is not specified, look for an available implementation 6278 // of the required type in effect factory 6279 if (EffectIsNullUuid(&pDesc->type)) { 6280 ALOGW("createEffect() no effect type"); 6281 lStatus = BAD_VALUE; 6282 goto Exit; 6283 } 6284 uint32_t numEffects = 0; 6285 effect_descriptor_t d; 6286 d.flags = 0; // prevent compiler warning 6287 bool found = false; 6288 6289 lStatus = EffectQueryNumberEffects(&numEffects); 6290 if (lStatus < 0) { 6291 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6292 goto Exit; 6293 } 6294 for (uint32_t i = 0; i < numEffects; i++) { 6295 lStatus = EffectQueryEffect(i, &desc); 6296 if (lStatus < 0) { 6297 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6298 continue; 6299 } 6300 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6301 // If matching type found save effect descriptor. If the session is 6302 // 0 and the effect is not auxiliary, continue enumeration in case 6303 // an auxiliary version of this effect type is available 6304 found = true; 6305 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6306 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6307 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6308 break; 6309 } 6310 } 6311 } 6312 if (!found) { 6313 lStatus = BAD_VALUE; 6314 ALOGW("createEffect() effect not found"); 6315 goto Exit; 6316 } 6317 // For same effect type, chose auxiliary version over insert version if 6318 // connect to output mix (Compliance to OpenSL ES) 6319 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6320 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6321 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6322 } 6323 } 6324 6325 // Do not allow auxiliary effects on a session different from 0 (output mix) 6326 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6327 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6328 lStatus = INVALID_OPERATION; 6329 goto Exit; 6330 } 6331 6332 // check recording permission for visualizer 6333 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6334 !recordingAllowed()) { 6335 lStatus = PERMISSION_DENIED; 6336 goto Exit; 6337 } 6338 6339 // return effect descriptor 6340 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6341 6342 // If output is not specified try to find a matching audio session ID in one of the 6343 // output threads. 6344 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6345 // because of code checking output when entering the function. 6346 // Note: io is never 0 when creating an effect on an input 6347 if (io == 0) { 6348 // look for the thread where the specified audio session is present 6349 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6350 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6351 io = mPlaybackThreads.keyAt(i); 6352 break; 6353 } 6354 } 6355 if (io == 0) { 6356 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6357 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6358 io = mRecordThreads.keyAt(i); 6359 break; 6360 } 6361 } 6362 } 6363 // If no output thread contains the requested session ID, default to 6364 // first output. The effect chain will be moved to the correct output 6365 // thread when a track with the same session ID is created 6366 if (io == 0 && mPlaybackThreads.size()) { 6367 io = mPlaybackThreads.keyAt(0); 6368 } 6369 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6370 } 6371 ThreadBase *thread = checkRecordThread_l(io); 6372 if (thread == NULL) { 6373 thread = checkPlaybackThread_l(io); 6374 if (thread == NULL) { 6375 ALOGE("createEffect() unknown output thread"); 6376 lStatus = BAD_VALUE; 6377 goto Exit; 6378 } 6379 } 6380 6381 sp<Client> client = registerPid_l(pid); 6382 6383 // create effect on selected output thread 6384 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6385 &desc, enabled, &lStatus); 6386 if (handle != 0 && id != NULL) { 6387 *id = handle->id(); 6388 } 6389 } 6390 6391Exit: 6392 if (status != NULL) { 6393 *status = lStatus; 6394 } 6395 return handle; 6396} 6397 6398status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6399 audio_io_handle_t dstOutput) 6400{ 6401 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6402 sessionId, srcOutput, dstOutput); 6403 Mutex::Autolock _l(mLock); 6404 if (srcOutput == dstOutput) { 6405 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6406 return NO_ERROR; 6407 } 6408 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6409 if (srcThread == NULL) { 6410 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6411 return BAD_VALUE; 6412 } 6413 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6414 if (dstThread == NULL) { 6415 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6416 return BAD_VALUE; 6417 } 6418 6419 Mutex::Autolock _dl(dstThread->mLock); 6420 Mutex::Autolock _sl(srcThread->mLock); 6421 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6422 6423 return NO_ERROR; 6424} 6425 6426// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6427status_t AudioFlinger::moveEffectChain_l(int sessionId, 6428 AudioFlinger::PlaybackThread *srcThread, 6429 AudioFlinger::PlaybackThread *dstThread, 6430 bool reRegister) 6431{ 6432 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6433 sessionId, srcThread, dstThread); 6434 6435 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6436 if (chain == 0) { 6437 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6438 sessionId, srcThread); 6439 return INVALID_OPERATION; 6440 } 6441 6442 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6443 // so that a new chain is created with correct parameters when first effect is added. This is 6444 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6445 // removed. 6446 srcThread->removeEffectChain_l(chain); 6447 6448 // transfer all effects one by one so that new effect chain is created on new thread with 6449 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6450 audio_io_handle_t dstOutput = dstThread->id(); 6451 sp<EffectChain> dstChain; 6452 uint32_t strategy = 0; // prevent compiler warning 6453 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6454 while (effect != 0) { 6455 srcThread->removeEffect_l(effect); 6456 dstThread->addEffect_l(effect); 6457 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6458 if (effect->state() == EffectModule::ACTIVE || 6459 effect->state() == EffectModule::STOPPING) { 6460 effect->start(); 6461 } 6462 // if the move request is not received from audio policy manager, the effect must be 6463 // re-registered with the new strategy and output 6464 if (dstChain == 0) { 6465 dstChain = effect->chain().promote(); 6466 if (dstChain == 0) { 6467 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6468 srcThread->addEffect_l(effect); 6469 return NO_INIT; 6470 } 6471 strategy = dstChain->strategy(); 6472 } 6473 if (reRegister) { 6474 AudioSystem::unregisterEffect(effect->id()); 6475 AudioSystem::registerEffect(&effect->desc(), 6476 dstOutput, 6477 strategy, 6478 sessionId, 6479 effect->id()); 6480 } 6481 effect = chain->getEffectFromId_l(0); 6482 } 6483 6484 return NO_ERROR; 6485} 6486 6487 6488// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6489sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6490 const sp<AudioFlinger::Client>& client, 6491 const sp<IEffectClient>& effectClient, 6492 int32_t priority, 6493 int sessionId, 6494 effect_descriptor_t *desc, 6495 int *enabled, 6496 status_t *status 6497 ) 6498{ 6499 sp<EffectModule> effect; 6500 sp<EffectHandle> handle; 6501 status_t lStatus; 6502 sp<EffectChain> chain; 6503 bool chainCreated = false; 6504 bool effectCreated = false; 6505 bool effectRegistered = false; 6506 6507 lStatus = initCheck(); 6508 if (lStatus != NO_ERROR) { 6509 ALOGW("createEffect_l() Audio driver not initialized."); 6510 goto Exit; 6511 } 6512 6513 // Do not allow effects with session ID 0 on direct output or duplicating threads 6514 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6515 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6516 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6517 desc->name, sessionId); 6518 lStatus = BAD_VALUE; 6519 goto Exit; 6520 } 6521 // Only Pre processor effects are allowed on input threads and only on input threads 6522 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6523 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6524 desc->name, desc->flags, mType); 6525 lStatus = BAD_VALUE; 6526 goto Exit; 6527 } 6528 6529 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6530 6531 { // scope for mLock 6532 Mutex::Autolock _l(mLock); 6533 6534 // check for existing effect chain with the requested audio session 6535 chain = getEffectChain_l(sessionId); 6536 if (chain == 0) { 6537 // create a new chain for this session 6538 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6539 chain = new EffectChain(this, sessionId); 6540 addEffectChain_l(chain); 6541 chain->setStrategy(getStrategyForSession_l(sessionId)); 6542 chainCreated = true; 6543 } else { 6544 effect = chain->getEffectFromDesc_l(desc); 6545 } 6546 6547 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6548 6549 if (effect == 0) { 6550 int id = mAudioFlinger->nextUniqueId(); 6551 // Check CPU and memory usage 6552 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6553 if (lStatus != NO_ERROR) { 6554 goto Exit; 6555 } 6556 effectRegistered = true; 6557 // create a new effect module if none present in the chain 6558 effect = new EffectModule(this, chain, desc, id, sessionId); 6559 lStatus = effect->status(); 6560 if (lStatus != NO_ERROR) { 6561 goto Exit; 6562 } 6563 lStatus = chain->addEffect_l(effect); 6564 if (lStatus != NO_ERROR) { 6565 goto Exit; 6566 } 6567 effectCreated = true; 6568 6569 effect->setDevice(mDevice); 6570 effect->setMode(mAudioFlinger->getMode()); 6571 } 6572 // create effect handle and connect it to effect module 6573 handle = new EffectHandle(effect, client, effectClient, priority); 6574 lStatus = effect->addHandle(handle); 6575 if (enabled != NULL) { 6576 *enabled = (int)effect->isEnabled(); 6577 } 6578 } 6579 6580Exit: 6581 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6582 Mutex::Autolock _l(mLock); 6583 if (effectCreated) { 6584 chain->removeEffect_l(effect); 6585 } 6586 if (effectRegistered) { 6587 AudioSystem::unregisterEffect(effect->id()); 6588 } 6589 if (chainCreated) { 6590 removeEffectChain_l(chain); 6591 } 6592 handle.clear(); 6593 } 6594 6595 if (status != NULL) { 6596 *status = lStatus; 6597 } 6598 return handle; 6599} 6600 6601sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6602{ 6603 sp<EffectChain> chain = getEffectChain_l(sessionId); 6604 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6605} 6606 6607// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6608// PlaybackThread::mLock held 6609status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6610{ 6611 // check for existing effect chain with the requested audio session 6612 int sessionId = effect->sessionId(); 6613 sp<EffectChain> chain = getEffectChain_l(sessionId); 6614 bool chainCreated = false; 6615 6616 if (chain == 0) { 6617 // create a new chain for this session 6618 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6619 chain = new EffectChain(this, sessionId); 6620 addEffectChain_l(chain); 6621 chain->setStrategy(getStrategyForSession_l(sessionId)); 6622 chainCreated = true; 6623 } 6624 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6625 6626 if (chain->getEffectFromId_l(effect->id()) != 0) { 6627 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6628 this, effect->desc().name, chain.get()); 6629 return BAD_VALUE; 6630 } 6631 6632 status_t status = chain->addEffect_l(effect); 6633 if (status != NO_ERROR) { 6634 if (chainCreated) { 6635 removeEffectChain_l(chain); 6636 } 6637 return status; 6638 } 6639 6640 effect->setDevice(mDevice); 6641 effect->setMode(mAudioFlinger->getMode()); 6642 return NO_ERROR; 6643} 6644 6645void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6646 6647 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6648 effect_descriptor_t desc = effect->desc(); 6649 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6650 detachAuxEffect_l(effect->id()); 6651 } 6652 6653 sp<EffectChain> chain = effect->chain().promote(); 6654 if (chain != 0) { 6655 // remove effect chain if removing last effect 6656 if (chain->removeEffect_l(effect) == 0) { 6657 removeEffectChain_l(chain); 6658 } 6659 } else { 6660 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6661 } 6662} 6663 6664void AudioFlinger::ThreadBase::lockEffectChains_l( 6665 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6666{ 6667 effectChains = mEffectChains; 6668 for (size_t i = 0; i < mEffectChains.size(); i++) { 6669 mEffectChains[i]->lock(); 6670 } 6671} 6672 6673void AudioFlinger::ThreadBase::unlockEffectChains( 6674 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6675{ 6676 for (size_t i = 0; i < effectChains.size(); i++) { 6677 effectChains[i]->unlock(); 6678 } 6679} 6680 6681sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6682{ 6683 Mutex::Autolock _l(mLock); 6684 return getEffectChain_l(sessionId); 6685} 6686 6687sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6688{ 6689 size_t size = mEffectChains.size(); 6690 for (size_t i = 0; i < size; i++) { 6691 if (mEffectChains[i]->sessionId() == sessionId) { 6692 return mEffectChains[i]; 6693 } 6694 } 6695 return 0; 6696} 6697 6698void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6699{ 6700 Mutex::Autolock _l(mLock); 6701 size_t size = mEffectChains.size(); 6702 for (size_t i = 0; i < size; i++) { 6703 mEffectChains[i]->setMode_l(mode); 6704 } 6705} 6706 6707void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6708 const wp<EffectHandle>& handle, 6709 bool unpinIfLast) { 6710 6711 Mutex::Autolock _l(mLock); 6712 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6713 // delete the effect module if removing last handle on it 6714 if (effect->removeHandle(handle) == 0) { 6715 if (!effect->isPinned() || unpinIfLast) { 6716 removeEffect_l(effect); 6717 AudioSystem::unregisterEffect(effect->id()); 6718 } 6719 } 6720} 6721 6722status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6723{ 6724 int session = chain->sessionId(); 6725 int16_t *buffer = mMixBuffer; 6726 bool ownsBuffer = false; 6727 6728 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6729 if (session > 0) { 6730 // Only one effect chain can be present in direct output thread and it uses 6731 // the mix buffer as input 6732 if (mType != DIRECT) { 6733 size_t numSamples = mFrameCount * mChannelCount; 6734 buffer = new int16_t[numSamples]; 6735 memset(buffer, 0, numSamples * sizeof(int16_t)); 6736 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6737 ownsBuffer = true; 6738 } 6739 6740 // Attach all tracks with same session ID to this chain. 6741 for (size_t i = 0; i < mTracks.size(); ++i) { 6742 sp<Track> track = mTracks[i]; 6743 if (session == track->sessionId()) { 6744 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6745 track->setMainBuffer(buffer); 6746 chain->incTrackCnt(); 6747 } 6748 } 6749 6750 // indicate all active tracks in the chain 6751 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6752 sp<Track> track = mActiveTracks[i].promote(); 6753 if (track == 0) continue; 6754 if (session == track->sessionId()) { 6755 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6756 chain->incActiveTrackCnt(); 6757 } 6758 } 6759 } 6760 6761 chain->setInBuffer(buffer, ownsBuffer); 6762 chain->setOutBuffer(mMixBuffer); 6763 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6764 // chains list in order to be processed last as it contains output stage effects 6765 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6766 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6767 // after track specific effects and before output stage 6768 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6769 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6770 // Effect chain for other sessions are inserted at beginning of effect 6771 // chains list to be processed before output mix effects. Relative order between other 6772 // sessions is not important 6773 size_t size = mEffectChains.size(); 6774 size_t i = 0; 6775 for (i = 0; i < size; i++) { 6776 if (mEffectChains[i]->sessionId() < session) break; 6777 } 6778 mEffectChains.insertAt(chain, i); 6779 checkSuspendOnAddEffectChain_l(chain); 6780 6781 return NO_ERROR; 6782} 6783 6784size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6785{ 6786 int session = chain->sessionId(); 6787 6788 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6789 6790 for (size_t i = 0; i < mEffectChains.size(); i++) { 6791 if (chain == mEffectChains[i]) { 6792 mEffectChains.removeAt(i); 6793 // detach all active tracks from the chain 6794 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6795 sp<Track> track = mActiveTracks[i].promote(); 6796 if (track == 0) continue; 6797 if (session == track->sessionId()) { 6798 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6799 chain.get(), session); 6800 chain->decActiveTrackCnt(); 6801 } 6802 } 6803 6804 // detach all tracks with same session ID from this chain 6805 for (size_t i = 0; i < mTracks.size(); ++i) { 6806 sp<Track> track = mTracks[i]; 6807 if (session == track->sessionId()) { 6808 track->setMainBuffer(mMixBuffer); 6809 chain->decTrackCnt(); 6810 } 6811 } 6812 break; 6813 } 6814 } 6815 return mEffectChains.size(); 6816} 6817 6818status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6819 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6820{ 6821 Mutex::Autolock _l(mLock); 6822 return attachAuxEffect_l(track, EffectId); 6823} 6824 6825status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6826 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6827{ 6828 status_t status = NO_ERROR; 6829 6830 if (EffectId == 0) { 6831 track->setAuxBuffer(0, NULL); 6832 } else { 6833 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6834 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6835 if (effect != 0) { 6836 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6837 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6838 } else { 6839 status = INVALID_OPERATION; 6840 } 6841 } else { 6842 status = BAD_VALUE; 6843 } 6844 } 6845 return status; 6846} 6847 6848void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6849{ 6850 for (size_t i = 0; i < mTracks.size(); ++i) { 6851 sp<Track> track = mTracks[i]; 6852 if (track->auxEffectId() == effectId) { 6853 attachAuxEffect_l(track, 0); 6854 } 6855 } 6856} 6857 6858status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6859{ 6860 // only one chain per input thread 6861 if (mEffectChains.size() != 0) { 6862 return INVALID_OPERATION; 6863 } 6864 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6865 6866 chain->setInBuffer(NULL); 6867 chain->setOutBuffer(NULL); 6868 6869 checkSuspendOnAddEffectChain_l(chain); 6870 6871 mEffectChains.add(chain); 6872 6873 return NO_ERROR; 6874} 6875 6876size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6877{ 6878 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6879 ALOGW_IF(mEffectChains.size() != 1, 6880 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6881 chain.get(), mEffectChains.size(), this); 6882 if (mEffectChains.size() == 1) { 6883 mEffectChains.removeAt(0); 6884 } 6885 return 0; 6886} 6887 6888// ---------------------------------------------------------------------------- 6889// EffectModule implementation 6890// ---------------------------------------------------------------------------- 6891 6892#undef LOG_TAG 6893#define LOG_TAG "AudioFlinger::EffectModule" 6894 6895AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6896 const wp<AudioFlinger::EffectChain>& chain, 6897 effect_descriptor_t *desc, 6898 int id, 6899 int sessionId) 6900 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6901 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6902{ 6903 ALOGV("Constructor %p", this); 6904 int lStatus; 6905 if (thread == NULL) { 6906 return; 6907 } 6908 6909 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6910 6911 // create effect engine from effect factory 6912 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6913 6914 if (mStatus != NO_ERROR) { 6915 return; 6916 } 6917 lStatus = init(); 6918 if (lStatus < 0) { 6919 mStatus = lStatus; 6920 goto Error; 6921 } 6922 6923 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6924 mPinned = true; 6925 } 6926 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6927 return; 6928Error: 6929 EffectRelease(mEffectInterface); 6930 mEffectInterface = NULL; 6931 ALOGV("Constructor Error %d", mStatus); 6932} 6933 6934AudioFlinger::EffectModule::~EffectModule() 6935{ 6936 ALOGV("Destructor %p", this); 6937 if (mEffectInterface != NULL) { 6938 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6939 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6940 sp<ThreadBase> thread = mThread.promote(); 6941 if (thread != 0) { 6942 audio_stream_t *stream = thread->stream(); 6943 if (stream != NULL) { 6944 stream->remove_audio_effect(stream, mEffectInterface); 6945 } 6946 } 6947 } 6948 // release effect engine 6949 EffectRelease(mEffectInterface); 6950 } 6951} 6952 6953status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6954{ 6955 status_t status; 6956 6957 Mutex::Autolock _l(mLock); 6958 int priority = handle->priority(); 6959 size_t size = mHandles.size(); 6960 sp<EffectHandle> h; 6961 size_t i; 6962 for (i = 0; i < size; i++) { 6963 h = mHandles[i].promote(); 6964 if (h == 0) continue; 6965 if (h->priority() <= priority) break; 6966 } 6967 // if inserted in first place, move effect control from previous owner to this handle 6968 if (i == 0) { 6969 bool enabled = false; 6970 if (h != 0) { 6971 enabled = h->enabled(); 6972 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6973 } 6974 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6975 status = NO_ERROR; 6976 } else { 6977 status = ALREADY_EXISTS; 6978 } 6979 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6980 mHandles.insertAt(handle, i); 6981 return status; 6982} 6983 6984size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6985{ 6986 Mutex::Autolock _l(mLock); 6987 size_t size = mHandles.size(); 6988 size_t i; 6989 for (i = 0; i < size; i++) { 6990 if (mHandles[i] == handle) break; 6991 } 6992 if (i == size) { 6993 return size; 6994 } 6995 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6996 6997 bool enabled = false; 6998 EffectHandle *hdl = handle.unsafe_get(); 6999 if (hdl != NULL) { 7000 ALOGV("removeHandle() unsafe_get OK"); 7001 enabled = hdl->enabled(); 7002 } 7003 mHandles.removeAt(i); 7004 size = mHandles.size(); 7005 // if removed from first place, move effect control from this handle to next in line 7006 if (i == 0 && size != 0) { 7007 sp<EffectHandle> h = mHandles[0].promote(); 7008 if (h != 0) { 7009 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7010 } 7011 } 7012 7013 // Prevent calls to process() and other functions on effect interface from now on. 7014 // The effect engine will be released by the destructor when the last strong reference on 7015 // this object is released which can happen after next process is called. 7016 if (size == 0 && !mPinned) { 7017 mState = DESTROYED; 7018 } 7019 7020 return size; 7021} 7022 7023sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7024{ 7025 Mutex::Autolock _l(mLock); 7026 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7027} 7028 7029void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7030{ 7031 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7032 // keep a strong reference on this EffectModule to avoid calling the 7033 // destructor before we exit 7034 sp<EffectModule> keep(this); 7035 { 7036 sp<ThreadBase> thread = mThread.promote(); 7037 if (thread != 0) { 7038 thread->disconnectEffect(keep, handle, unpinIfLast); 7039 } 7040 } 7041} 7042 7043void AudioFlinger::EffectModule::updateState() { 7044 Mutex::Autolock _l(mLock); 7045 7046 switch (mState) { 7047 case RESTART: 7048 reset_l(); 7049 // FALL THROUGH 7050 7051 case STARTING: 7052 // clear auxiliary effect input buffer for next accumulation 7053 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7054 memset(mConfig.inputCfg.buffer.raw, 7055 0, 7056 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7057 } 7058 start_l(); 7059 mState = ACTIVE; 7060 break; 7061 case STOPPING: 7062 stop_l(); 7063 mDisableWaitCnt = mMaxDisableWaitCnt; 7064 mState = STOPPED; 7065 break; 7066 case STOPPED: 7067 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7068 // turn off sequence. 7069 if (--mDisableWaitCnt == 0) { 7070 reset_l(); 7071 mState = IDLE; 7072 } 7073 break; 7074 default: //IDLE , ACTIVE, DESTROYED 7075 break; 7076 } 7077} 7078 7079void AudioFlinger::EffectModule::process() 7080{ 7081 Mutex::Autolock _l(mLock); 7082 7083 if (mState == DESTROYED || mEffectInterface == NULL || 7084 mConfig.inputCfg.buffer.raw == NULL || 7085 mConfig.outputCfg.buffer.raw == NULL) { 7086 return; 7087 } 7088 7089 if (isProcessEnabled()) { 7090 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7091 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7092 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7093 mConfig.inputCfg.buffer.s32, 7094 mConfig.inputCfg.buffer.frameCount/2); 7095 } 7096 7097 // do the actual processing in the effect engine 7098 int ret = (*mEffectInterface)->process(mEffectInterface, 7099 &mConfig.inputCfg.buffer, 7100 &mConfig.outputCfg.buffer); 7101 7102 // force transition to IDLE state when engine is ready 7103 if (mState == STOPPED && ret == -ENODATA) { 7104 mDisableWaitCnt = 1; 7105 } 7106 7107 // clear auxiliary effect input buffer for next accumulation 7108 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7109 memset(mConfig.inputCfg.buffer.raw, 0, 7110 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7111 } 7112 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7113 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7114 // If an insert effect is idle and input buffer is different from output buffer, 7115 // accumulate input onto output 7116 sp<EffectChain> chain = mChain.promote(); 7117 if (chain != 0 && chain->activeTrackCnt() != 0) { 7118 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7119 int16_t *in = mConfig.inputCfg.buffer.s16; 7120 int16_t *out = mConfig.outputCfg.buffer.s16; 7121 for (size_t i = 0; i < frameCnt; i++) { 7122 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7123 } 7124 } 7125 } 7126} 7127 7128void AudioFlinger::EffectModule::reset_l() 7129{ 7130 if (mEffectInterface == NULL) { 7131 return; 7132 } 7133 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7134} 7135 7136status_t AudioFlinger::EffectModule::configure() 7137{ 7138 uint32_t channels; 7139 if (mEffectInterface == NULL) { 7140 return NO_INIT; 7141 } 7142 7143 sp<ThreadBase> thread = mThread.promote(); 7144 if (thread == 0) { 7145 return DEAD_OBJECT; 7146 } 7147 7148 // TODO: handle configuration of effects replacing track process 7149 if (thread->channelCount() == 1) { 7150 channels = AUDIO_CHANNEL_OUT_MONO; 7151 } else { 7152 channels = AUDIO_CHANNEL_OUT_STEREO; 7153 } 7154 7155 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7156 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7157 } else { 7158 mConfig.inputCfg.channels = channels; 7159 } 7160 mConfig.outputCfg.channels = channels; 7161 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7162 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7163 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7164 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7165 mConfig.inputCfg.bufferProvider.cookie = NULL; 7166 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7167 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7168 mConfig.outputCfg.bufferProvider.cookie = NULL; 7169 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7170 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7171 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7172 // Insert effect: 7173 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7174 // always overwrites output buffer: input buffer == output buffer 7175 // - in other sessions: 7176 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7177 // other effect: overwrites output buffer: input buffer == output buffer 7178 // Auxiliary effect: 7179 // accumulates in output buffer: input buffer != output buffer 7180 // Therefore: accumulate <=> input buffer != output buffer 7181 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7182 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7183 } else { 7184 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7185 } 7186 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7187 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7188 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7189 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7190 7191 ALOGV("configure() %p thread %p buffer %p framecount %d", 7192 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7193 7194 status_t cmdStatus; 7195 uint32_t size = sizeof(int); 7196 status_t status = (*mEffectInterface)->command(mEffectInterface, 7197 EFFECT_CMD_SET_CONFIG, 7198 sizeof(effect_config_t), 7199 &mConfig, 7200 &size, 7201 &cmdStatus); 7202 if (status == 0) { 7203 status = cmdStatus; 7204 } 7205 7206 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7207 (1000 * mConfig.outputCfg.buffer.frameCount); 7208 7209 return status; 7210} 7211 7212status_t AudioFlinger::EffectModule::init() 7213{ 7214 Mutex::Autolock _l(mLock); 7215 if (mEffectInterface == NULL) { 7216 return NO_INIT; 7217 } 7218 status_t cmdStatus; 7219 uint32_t size = sizeof(status_t); 7220 status_t status = (*mEffectInterface)->command(mEffectInterface, 7221 EFFECT_CMD_INIT, 7222 0, 7223 NULL, 7224 &size, 7225 &cmdStatus); 7226 if (status == 0) { 7227 status = cmdStatus; 7228 } 7229 return status; 7230} 7231 7232status_t AudioFlinger::EffectModule::start() 7233{ 7234 Mutex::Autolock _l(mLock); 7235 return start_l(); 7236} 7237 7238status_t AudioFlinger::EffectModule::start_l() 7239{ 7240 if (mEffectInterface == NULL) { 7241 return NO_INIT; 7242 } 7243 status_t cmdStatus; 7244 uint32_t size = sizeof(status_t); 7245 status_t status = (*mEffectInterface)->command(mEffectInterface, 7246 EFFECT_CMD_ENABLE, 7247 0, 7248 NULL, 7249 &size, 7250 &cmdStatus); 7251 if (status == 0) { 7252 status = cmdStatus; 7253 } 7254 if (status == 0 && 7255 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7256 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7257 sp<ThreadBase> thread = mThread.promote(); 7258 if (thread != 0) { 7259 audio_stream_t *stream = thread->stream(); 7260 if (stream != NULL) { 7261 stream->add_audio_effect(stream, mEffectInterface); 7262 } 7263 } 7264 } 7265 return status; 7266} 7267 7268status_t AudioFlinger::EffectModule::stop() 7269{ 7270 Mutex::Autolock _l(mLock); 7271 return stop_l(); 7272} 7273 7274status_t AudioFlinger::EffectModule::stop_l() 7275{ 7276 if (mEffectInterface == NULL) { 7277 return NO_INIT; 7278 } 7279 status_t cmdStatus; 7280 uint32_t size = sizeof(status_t); 7281 status_t status = (*mEffectInterface)->command(mEffectInterface, 7282 EFFECT_CMD_DISABLE, 7283 0, 7284 NULL, 7285 &size, 7286 &cmdStatus); 7287 if (status == 0) { 7288 status = cmdStatus; 7289 } 7290 if (status == 0 && 7291 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7292 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7293 sp<ThreadBase> thread = mThread.promote(); 7294 if (thread != 0) { 7295 audio_stream_t *stream = thread->stream(); 7296 if (stream != NULL) { 7297 stream->remove_audio_effect(stream, mEffectInterface); 7298 } 7299 } 7300 } 7301 return status; 7302} 7303 7304status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7305 uint32_t cmdSize, 7306 void *pCmdData, 7307 uint32_t *replySize, 7308 void *pReplyData) 7309{ 7310 Mutex::Autolock _l(mLock); 7311// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7312 7313 if (mState == DESTROYED || mEffectInterface == NULL) { 7314 return NO_INIT; 7315 } 7316 status_t status = (*mEffectInterface)->command(mEffectInterface, 7317 cmdCode, 7318 cmdSize, 7319 pCmdData, 7320 replySize, 7321 pReplyData); 7322 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7323 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7324 for (size_t i = 1; i < mHandles.size(); i++) { 7325 sp<EffectHandle> h = mHandles[i].promote(); 7326 if (h != 0) { 7327 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7328 } 7329 } 7330 } 7331 return status; 7332} 7333 7334status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7335{ 7336 7337 Mutex::Autolock _l(mLock); 7338 ALOGV("setEnabled %p enabled %d", this, enabled); 7339 7340 if (enabled != isEnabled()) { 7341 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7342 if (enabled && status != NO_ERROR) { 7343 return status; 7344 } 7345 7346 switch (mState) { 7347 // going from disabled to enabled 7348 case IDLE: 7349 mState = STARTING; 7350 break; 7351 case STOPPED: 7352 mState = RESTART; 7353 break; 7354 case STOPPING: 7355 mState = ACTIVE; 7356 break; 7357 7358 // going from enabled to disabled 7359 case RESTART: 7360 mState = STOPPED; 7361 break; 7362 case STARTING: 7363 mState = IDLE; 7364 break; 7365 case ACTIVE: 7366 mState = STOPPING; 7367 break; 7368 case DESTROYED: 7369 return NO_ERROR; // simply ignore as we are being destroyed 7370 } 7371 for (size_t i = 1; i < mHandles.size(); i++) { 7372 sp<EffectHandle> h = mHandles[i].promote(); 7373 if (h != 0) { 7374 h->setEnabled(enabled); 7375 } 7376 } 7377 } 7378 return NO_ERROR; 7379} 7380 7381bool AudioFlinger::EffectModule::isEnabled() const 7382{ 7383 switch (mState) { 7384 case RESTART: 7385 case STARTING: 7386 case ACTIVE: 7387 return true; 7388 case IDLE: 7389 case STOPPING: 7390 case STOPPED: 7391 case DESTROYED: 7392 default: 7393 return false; 7394 } 7395} 7396 7397bool AudioFlinger::EffectModule::isProcessEnabled() const 7398{ 7399 switch (mState) { 7400 case RESTART: 7401 case ACTIVE: 7402 case STOPPING: 7403 case STOPPED: 7404 return true; 7405 case IDLE: 7406 case STARTING: 7407 case DESTROYED: 7408 default: 7409 return false; 7410 } 7411} 7412 7413status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7414{ 7415 Mutex::Autolock _l(mLock); 7416 status_t status = NO_ERROR; 7417 7418 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7419 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7420 if (isProcessEnabled() && 7421 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7422 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7423 status_t cmdStatus; 7424 uint32_t volume[2]; 7425 uint32_t *pVolume = NULL; 7426 uint32_t size = sizeof(volume); 7427 volume[0] = *left; 7428 volume[1] = *right; 7429 if (controller) { 7430 pVolume = volume; 7431 } 7432 status = (*mEffectInterface)->command(mEffectInterface, 7433 EFFECT_CMD_SET_VOLUME, 7434 size, 7435 volume, 7436 &size, 7437 pVolume); 7438 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7439 *left = volume[0]; 7440 *right = volume[1]; 7441 } 7442 } 7443 return status; 7444} 7445 7446status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7447{ 7448 Mutex::Autolock _l(mLock); 7449 status_t status = NO_ERROR; 7450 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7451 // audio pre processing modules on RecordThread can receive both output and 7452 // input device indication in the same call 7453 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7454 if (dev) { 7455 status_t cmdStatus; 7456 uint32_t size = sizeof(status_t); 7457 7458 status = (*mEffectInterface)->command(mEffectInterface, 7459 EFFECT_CMD_SET_DEVICE, 7460 sizeof(uint32_t), 7461 &dev, 7462 &size, 7463 &cmdStatus); 7464 if (status == NO_ERROR) { 7465 status = cmdStatus; 7466 } 7467 } 7468 dev = device & AUDIO_DEVICE_IN_ALL; 7469 if (dev) { 7470 status_t cmdStatus; 7471 uint32_t size = sizeof(status_t); 7472 7473 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7474 EFFECT_CMD_SET_INPUT_DEVICE, 7475 sizeof(uint32_t), 7476 &dev, 7477 &size, 7478 &cmdStatus); 7479 if (status2 == NO_ERROR) { 7480 status2 = cmdStatus; 7481 } 7482 if (status == NO_ERROR) { 7483 status = status2; 7484 } 7485 } 7486 } 7487 return status; 7488} 7489 7490status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7491{ 7492 Mutex::Autolock _l(mLock); 7493 status_t status = NO_ERROR; 7494 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7495 status_t cmdStatus; 7496 uint32_t size = sizeof(status_t); 7497 status = (*mEffectInterface)->command(mEffectInterface, 7498 EFFECT_CMD_SET_AUDIO_MODE, 7499 sizeof(audio_mode_t), 7500 &mode, 7501 &size, 7502 &cmdStatus); 7503 if (status == NO_ERROR) { 7504 status = cmdStatus; 7505 } 7506 } 7507 return status; 7508} 7509 7510void AudioFlinger::EffectModule::setSuspended(bool suspended) 7511{ 7512 Mutex::Autolock _l(mLock); 7513 mSuspended = suspended; 7514} 7515 7516bool AudioFlinger::EffectModule::suspended() const 7517{ 7518 Mutex::Autolock _l(mLock); 7519 return mSuspended; 7520} 7521 7522status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7523{ 7524 const size_t SIZE = 256; 7525 char buffer[SIZE]; 7526 String8 result; 7527 7528 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7529 result.append(buffer); 7530 7531 bool locked = tryLock(mLock); 7532 // failed to lock - AudioFlinger is probably deadlocked 7533 if (!locked) { 7534 result.append("\t\tCould not lock Fx mutex:\n"); 7535 } 7536 7537 result.append("\t\tSession Status State Engine:\n"); 7538 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7539 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7540 result.append(buffer); 7541 7542 result.append("\t\tDescriptor:\n"); 7543 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7544 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7545 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7546 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7547 result.append(buffer); 7548 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7549 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7550 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7551 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7552 result.append(buffer); 7553 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7554 mDescriptor.apiVersion, 7555 mDescriptor.flags); 7556 result.append(buffer); 7557 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7558 mDescriptor.name); 7559 result.append(buffer); 7560 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7561 mDescriptor.implementor); 7562 result.append(buffer); 7563 7564 result.append("\t\t- Input configuration:\n"); 7565 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7566 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7567 (uint32_t)mConfig.inputCfg.buffer.raw, 7568 mConfig.inputCfg.buffer.frameCount, 7569 mConfig.inputCfg.samplingRate, 7570 mConfig.inputCfg.channels, 7571 mConfig.inputCfg.format); 7572 result.append(buffer); 7573 7574 result.append("\t\t- Output configuration:\n"); 7575 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7576 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7577 (uint32_t)mConfig.outputCfg.buffer.raw, 7578 mConfig.outputCfg.buffer.frameCount, 7579 mConfig.outputCfg.samplingRate, 7580 mConfig.outputCfg.channels, 7581 mConfig.outputCfg.format); 7582 result.append(buffer); 7583 7584 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7585 result.append(buffer); 7586 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7587 for (size_t i = 0; i < mHandles.size(); ++i) { 7588 sp<EffectHandle> handle = mHandles[i].promote(); 7589 if (handle != 0) { 7590 handle->dump(buffer, SIZE); 7591 result.append(buffer); 7592 } 7593 } 7594 7595 result.append("\n"); 7596 7597 write(fd, result.string(), result.length()); 7598 7599 if (locked) { 7600 mLock.unlock(); 7601 } 7602 7603 return NO_ERROR; 7604} 7605 7606// ---------------------------------------------------------------------------- 7607// EffectHandle implementation 7608// ---------------------------------------------------------------------------- 7609 7610#undef LOG_TAG 7611#define LOG_TAG "AudioFlinger::EffectHandle" 7612 7613AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7614 const sp<AudioFlinger::Client>& client, 7615 const sp<IEffectClient>& effectClient, 7616 int32_t priority) 7617 : BnEffect(), 7618 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7619 mPriority(priority), mHasControl(false), mEnabled(false) 7620{ 7621 ALOGV("constructor %p", this); 7622 7623 if (client == 0) { 7624 return; 7625 } 7626 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7627 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7628 if (mCblkMemory != 0) { 7629 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7630 7631 if (mCblk != NULL) { 7632 new(mCblk) effect_param_cblk_t(); 7633 mBuffer = (uint8_t *)mCblk + bufOffset; 7634 } 7635 } else { 7636 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7637 return; 7638 } 7639} 7640 7641AudioFlinger::EffectHandle::~EffectHandle() 7642{ 7643 ALOGV("Destructor %p", this); 7644 disconnect(false); 7645 ALOGV("Destructor DONE %p", this); 7646} 7647 7648status_t AudioFlinger::EffectHandle::enable() 7649{ 7650 ALOGV("enable %p", this); 7651 if (!mHasControl) return INVALID_OPERATION; 7652 if (mEffect == 0) return DEAD_OBJECT; 7653 7654 if (mEnabled) { 7655 return NO_ERROR; 7656 } 7657 7658 mEnabled = true; 7659 7660 sp<ThreadBase> thread = mEffect->thread().promote(); 7661 if (thread != 0) { 7662 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7663 } 7664 7665 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7666 if (mEffect->suspended()) { 7667 return NO_ERROR; 7668 } 7669 7670 status_t status = mEffect->setEnabled(true); 7671 if (status != NO_ERROR) { 7672 if (thread != 0) { 7673 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7674 } 7675 mEnabled = false; 7676 } 7677 return status; 7678} 7679 7680status_t AudioFlinger::EffectHandle::disable() 7681{ 7682 ALOGV("disable %p", this); 7683 if (!mHasControl) return INVALID_OPERATION; 7684 if (mEffect == 0) return DEAD_OBJECT; 7685 7686 if (!mEnabled) { 7687 return NO_ERROR; 7688 } 7689 mEnabled = false; 7690 7691 if (mEffect->suspended()) { 7692 return NO_ERROR; 7693 } 7694 7695 status_t status = mEffect->setEnabled(false); 7696 7697 sp<ThreadBase> thread = mEffect->thread().promote(); 7698 if (thread != 0) { 7699 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7700 } 7701 7702 return status; 7703} 7704 7705void AudioFlinger::EffectHandle::disconnect() 7706{ 7707 disconnect(true); 7708} 7709 7710void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7711{ 7712 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7713 if (mEffect == 0) { 7714 return; 7715 } 7716 mEffect->disconnect(this, unpinIfLast); 7717 7718 if (mHasControl && mEnabled) { 7719 sp<ThreadBase> thread = mEffect->thread().promote(); 7720 if (thread != 0) { 7721 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7722 } 7723 } 7724 7725 // release sp on module => module destructor can be called now 7726 mEffect.clear(); 7727 if (mClient != 0) { 7728 if (mCblk != NULL) { 7729 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7730 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7731 } 7732 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7733 // Client destructor must run with AudioFlinger mutex locked 7734 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7735 mClient.clear(); 7736 } 7737} 7738 7739status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7740 uint32_t cmdSize, 7741 void *pCmdData, 7742 uint32_t *replySize, 7743 void *pReplyData) 7744{ 7745// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7746// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7747 7748 // only get parameter command is permitted for applications not controlling the effect 7749 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7750 return INVALID_OPERATION; 7751 } 7752 if (mEffect == 0) return DEAD_OBJECT; 7753 if (mClient == 0) return INVALID_OPERATION; 7754 7755 // handle commands that are not forwarded transparently to effect engine 7756 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7757 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7758 // no risk to block the whole media server process or mixer threads is we are stuck here 7759 Mutex::Autolock _l(mCblk->lock); 7760 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7761 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7762 mCblk->serverIndex = 0; 7763 mCblk->clientIndex = 0; 7764 return BAD_VALUE; 7765 } 7766 status_t status = NO_ERROR; 7767 while (mCblk->serverIndex < mCblk->clientIndex) { 7768 int reply; 7769 uint32_t rsize = sizeof(int); 7770 int *p = (int *)(mBuffer + mCblk->serverIndex); 7771 int size = *p++; 7772 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7773 ALOGW("command(): invalid parameter block size"); 7774 break; 7775 } 7776 effect_param_t *param = (effect_param_t *)p; 7777 if (param->psize == 0 || param->vsize == 0) { 7778 ALOGW("command(): null parameter or value size"); 7779 mCblk->serverIndex += size; 7780 continue; 7781 } 7782 uint32_t psize = sizeof(effect_param_t) + 7783 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7784 param->vsize; 7785 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7786 psize, 7787 p, 7788 &rsize, 7789 &reply); 7790 // stop at first error encountered 7791 if (ret != NO_ERROR) { 7792 status = ret; 7793 *(int *)pReplyData = reply; 7794 break; 7795 } else if (reply != NO_ERROR) { 7796 *(int *)pReplyData = reply; 7797 break; 7798 } 7799 mCblk->serverIndex += size; 7800 } 7801 mCblk->serverIndex = 0; 7802 mCblk->clientIndex = 0; 7803 return status; 7804 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7805 *(int *)pReplyData = NO_ERROR; 7806 return enable(); 7807 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7808 *(int *)pReplyData = NO_ERROR; 7809 return disable(); 7810 } 7811 7812 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7813} 7814 7815void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7816{ 7817 ALOGV("setControl %p control %d", this, hasControl); 7818 7819 mHasControl = hasControl; 7820 mEnabled = enabled; 7821 7822 if (signal && mEffectClient != 0) { 7823 mEffectClient->controlStatusChanged(hasControl); 7824 } 7825} 7826 7827void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7828 uint32_t cmdSize, 7829 void *pCmdData, 7830 uint32_t replySize, 7831 void *pReplyData) 7832{ 7833 if (mEffectClient != 0) { 7834 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7835 } 7836} 7837 7838 7839 7840void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7841{ 7842 if (mEffectClient != 0) { 7843 mEffectClient->enableStatusChanged(enabled); 7844 } 7845} 7846 7847status_t AudioFlinger::EffectHandle::onTransact( 7848 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7849{ 7850 return BnEffect::onTransact(code, data, reply, flags); 7851} 7852 7853 7854void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7855{ 7856 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7857 7858 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7859 (mClient == 0) ? getpid_cached : mClient->pid(), 7860 mPriority, 7861 mHasControl, 7862 !locked, 7863 mCblk ? mCblk->clientIndex : 0, 7864 mCblk ? mCblk->serverIndex : 0 7865 ); 7866 7867 if (locked) { 7868 mCblk->lock.unlock(); 7869 } 7870} 7871 7872#undef LOG_TAG 7873#define LOG_TAG "AudioFlinger::EffectChain" 7874 7875AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7876 int sessionId) 7877 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7878 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7879 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7880{ 7881 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7882 if (thread == NULL) { 7883 return; 7884 } 7885 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7886 thread->frameCount(); 7887} 7888 7889AudioFlinger::EffectChain::~EffectChain() 7890{ 7891 if (mOwnInBuffer) { 7892 delete mInBuffer; 7893 } 7894 7895} 7896 7897// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7898sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7899{ 7900 size_t size = mEffects.size(); 7901 7902 for (size_t i = 0; i < size; i++) { 7903 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7904 return mEffects[i]; 7905 } 7906 } 7907 return 0; 7908} 7909 7910// getEffectFromId_l() must be called with ThreadBase::mLock held 7911sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7912{ 7913 size_t size = mEffects.size(); 7914 7915 for (size_t i = 0; i < size; i++) { 7916 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7917 if (id == 0 || mEffects[i]->id() == id) { 7918 return mEffects[i]; 7919 } 7920 } 7921 return 0; 7922} 7923 7924// getEffectFromType_l() must be called with ThreadBase::mLock held 7925sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7926 const effect_uuid_t *type) 7927{ 7928 size_t size = mEffects.size(); 7929 7930 for (size_t i = 0; i < size; i++) { 7931 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7932 return mEffects[i]; 7933 } 7934 } 7935 return 0; 7936} 7937 7938// Must be called with EffectChain::mLock locked 7939void AudioFlinger::EffectChain::process_l() 7940{ 7941 sp<ThreadBase> thread = mThread.promote(); 7942 if (thread == 0) { 7943 ALOGW("process_l(): cannot promote mixer thread"); 7944 return; 7945 } 7946 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7947 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7948 // always process effects unless no more tracks are on the session and the effect tail 7949 // has been rendered 7950 bool doProcess = true; 7951 if (!isGlobalSession) { 7952 bool tracksOnSession = (trackCnt() != 0); 7953 7954 if (!tracksOnSession && mTailBufferCount == 0) { 7955 doProcess = false; 7956 } 7957 7958 if (activeTrackCnt() == 0) { 7959 // if no track is active and the effect tail has not been rendered, 7960 // the input buffer must be cleared here as the mixer process will not do it 7961 if (tracksOnSession || mTailBufferCount > 0) { 7962 size_t numSamples = thread->frameCount() * thread->channelCount(); 7963 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7964 if (mTailBufferCount > 0) { 7965 mTailBufferCount--; 7966 } 7967 } 7968 } 7969 } 7970 7971 size_t size = mEffects.size(); 7972 if (doProcess) { 7973 for (size_t i = 0; i < size; i++) { 7974 mEffects[i]->process(); 7975 } 7976 } 7977 for (size_t i = 0; i < size; i++) { 7978 mEffects[i]->updateState(); 7979 } 7980} 7981 7982// addEffect_l() must be called with PlaybackThread::mLock held 7983status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7984{ 7985 effect_descriptor_t desc = effect->desc(); 7986 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7987 7988 Mutex::Autolock _l(mLock); 7989 effect->setChain(this); 7990 sp<ThreadBase> thread = mThread.promote(); 7991 if (thread == 0) { 7992 return NO_INIT; 7993 } 7994 effect->setThread(thread); 7995 7996 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7997 // Auxiliary effects are inserted at the beginning of mEffects vector as 7998 // they are processed first and accumulated in chain input buffer 7999 mEffects.insertAt(effect, 0); 8000 8001 // the input buffer for auxiliary effect contains mono samples in 8002 // 32 bit format. This is to avoid saturation in AudoMixer 8003 // accumulation stage. Saturation is done in EffectModule::process() before 8004 // calling the process in effect engine 8005 size_t numSamples = thread->frameCount(); 8006 int32_t *buffer = new int32_t[numSamples]; 8007 memset(buffer, 0, numSamples * sizeof(int32_t)); 8008 effect->setInBuffer((int16_t *)buffer); 8009 // auxiliary effects output samples to chain input buffer for further processing 8010 // by insert effects 8011 effect->setOutBuffer(mInBuffer); 8012 } else { 8013 // Insert effects are inserted at the end of mEffects vector as they are processed 8014 // after track and auxiliary effects. 8015 // Insert effect order as a function of indicated preference: 8016 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8017 // another effect is present 8018 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8019 // last effect claiming first position 8020 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8021 // first effect claiming last position 8022 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8023 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8024 // already present 8025 8026 size_t size = mEffects.size(); 8027 size_t idx_insert = size; 8028 ssize_t idx_insert_first = -1; 8029 ssize_t idx_insert_last = -1; 8030 8031 for (size_t i = 0; i < size; i++) { 8032 effect_descriptor_t d = mEffects[i]->desc(); 8033 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8034 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8035 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8036 // check invalid effect chaining combinations 8037 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8038 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8039 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8040 return INVALID_OPERATION; 8041 } 8042 // remember position of first insert effect and by default 8043 // select this as insert position for new effect 8044 if (idx_insert == size) { 8045 idx_insert = i; 8046 } 8047 // remember position of last insert effect claiming 8048 // first position 8049 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8050 idx_insert_first = i; 8051 } 8052 // remember position of first insert effect claiming 8053 // last position 8054 if (iPref == EFFECT_FLAG_INSERT_LAST && 8055 idx_insert_last == -1) { 8056 idx_insert_last = i; 8057 } 8058 } 8059 } 8060 8061 // modify idx_insert from first position if needed 8062 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8063 if (idx_insert_last != -1) { 8064 idx_insert = idx_insert_last; 8065 } else { 8066 idx_insert = size; 8067 } 8068 } else { 8069 if (idx_insert_first != -1) { 8070 idx_insert = idx_insert_first + 1; 8071 } 8072 } 8073 8074 // always read samples from chain input buffer 8075 effect->setInBuffer(mInBuffer); 8076 8077 // if last effect in the chain, output samples to chain 8078 // output buffer, otherwise to chain input buffer 8079 if (idx_insert == size) { 8080 if (idx_insert != 0) { 8081 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8082 mEffects[idx_insert-1]->configure(); 8083 } 8084 effect->setOutBuffer(mOutBuffer); 8085 } else { 8086 effect->setOutBuffer(mInBuffer); 8087 } 8088 mEffects.insertAt(effect, idx_insert); 8089 8090 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8091 } 8092 effect->configure(); 8093 return NO_ERROR; 8094} 8095 8096// removeEffect_l() must be called with PlaybackThread::mLock held 8097size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8098{ 8099 Mutex::Autolock _l(mLock); 8100 size_t size = mEffects.size(); 8101 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8102 8103 for (size_t i = 0; i < size; i++) { 8104 if (effect == mEffects[i]) { 8105 // calling stop here will remove pre-processing effect from the audio HAL. 8106 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8107 // the middle of a read from audio HAL 8108 if (mEffects[i]->state() == EffectModule::ACTIVE || 8109 mEffects[i]->state() == EffectModule::STOPPING) { 8110 mEffects[i]->stop(); 8111 } 8112 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8113 delete[] effect->inBuffer(); 8114 } else { 8115 if (i == size - 1 && i != 0) { 8116 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8117 mEffects[i - 1]->configure(); 8118 } 8119 } 8120 mEffects.removeAt(i); 8121 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8122 break; 8123 } 8124 } 8125 8126 return mEffects.size(); 8127} 8128 8129// setDevice_l() must be called with PlaybackThread::mLock held 8130void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8131{ 8132 size_t size = mEffects.size(); 8133 for (size_t i = 0; i < size; i++) { 8134 mEffects[i]->setDevice(device); 8135 } 8136} 8137 8138// setMode_l() must be called with PlaybackThread::mLock held 8139void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8140{ 8141 size_t size = mEffects.size(); 8142 for (size_t i = 0; i < size; i++) { 8143 mEffects[i]->setMode(mode); 8144 } 8145} 8146 8147// setVolume_l() must be called with PlaybackThread::mLock held 8148bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8149{ 8150 uint32_t newLeft = *left; 8151 uint32_t newRight = *right; 8152 bool hasControl = false; 8153 int ctrlIdx = -1; 8154 size_t size = mEffects.size(); 8155 8156 // first update volume controller 8157 for (size_t i = size; i > 0; i--) { 8158 if (mEffects[i - 1]->isProcessEnabled() && 8159 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8160 ctrlIdx = i - 1; 8161 hasControl = true; 8162 break; 8163 } 8164 } 8165 8166 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8167 if (hasControl) { 8168 *left = mNewLeftVolume; 8169 *right = mNewRightVolume; 8170 } 8171 return hasControl; 8172 } 8173 8174 mVolumeCtrlIdx = ctrlIdx; 8175 mLeftVolume = newLeft; 8176 mRightVolume = newRight; 8177 8178 // second get volume update from volume controller 8179 if (ctrlIdx >= 0) { 8180 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8181 mNewLeftVolume = newLeft; 8182 mNewRightVolume = newRight; 8183 } 8184 // then indicate volume to all other effects in chain. 8185 // Pass altered volume to effects before volume controller 8186 // and requested volume to effects after controller 8187 uint32_t lVol = newLeft; 8188 uint32_t rVol = newRight; 8189 8190 for (size_t i = 0; i < size; i++) { 8191 if ((int)i == ctrlIdx) continue; 8192 // this also works for ctrlIdx == -1 when there is no volume controller 8193 if ((int)i > ctrlIdx) { 8194 lVol = *left; 8195 rVol = *right; 8196 } 8197 mEffects[i]->setVolume(&lVol, &rVol, false); 8198 } 8199 *left = newLeft; 8200 *right = newRight; 8201 8202 return hasControl; 8203} 8204 8205status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8206{ 8207 const size_t SIZE = 256; 8208 char buffer[SIZE]; 8209 String8 result; 8210 8211 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8212 result.append(buffer); 8213 8214 bool locked = tryLock(mLock); 8215 // failed to lock - AudioFlinger is probably deadlocked 8216 if (!locked) { 8217 result.append("\tCould not lock mutex:\n"); 8218 } 8219 8220 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8221 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8222 mEffects.size(), 8223 (uint32_t)mInBuffer, 8224 (uint32_t)mOutBuffer, 8225 mActiveTrackCnt); 8226 result.append(buffer); 8227 write(fd, result.string(), result.size()); 8228 8229 for (size_t i = 0; i < mEffects.size(); ++i) { 8230 sp<EffectModule> effect = mEffects[i]; 8231 if (effect != 0) { 8232 effect->dump(fd, args); 8233 } 8234 } 8235 8236 if (locked) { 8237 mLock.unlock(); 8238 } 8239 8240 return NO_ERROR; 8241} 8242 8243// must be called with ThreadBase::mLock held 8244void AudioFlinger::EffectChain::setEffectSuspended_l( 8245 const effect_uuid_t *type, bool suspend) 8246{ 8247 sp<SuspendedEffectDesc> desc; 8248 // use effect type UUID timelow as key as there is no real risk of identical 8249 // timeLow fields among effect type UUIDs. 8250 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8251 if (suspend) { 8252 if (index >= 0) { 8253 desc = mSuspendedEffects.valueAt(index); 8254 } else { 8255 desc = new SuspendedEffectDesc(); 8256 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8257 mSuspendedEffects.add(type->timeLow, desc); 8258 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8259 } 8260 if (desc->mRefCount++ == 0) { 8261 sp<EffectModule> effect = getEffectIfEnabled(type); 8262 if (effect != 0) { 8263 desc->mEffect = effect; 8264 effect->setSuspended(true); 8265 effect->setEnabled(false); 8266 } 8267 } 8268 } else { 8269 if (index < 0) { 8270 return; 8271 } 8272 desc = mSuspendedEffects.valueAt(index); 8273 if (desc->mRefCount <= 0) { 8274 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8275 desc->mRefCount = 1; 8276 } 8277 if (--desc->mRefCount == 0) { 8278 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8279 if (desc->mEffect != 0) { 8280 sp<EffectModule> effect = desc->mEffect.promote(); 8281 if (effect != 0) { 8282 effect->setSuspended(false); 8283 sp<EffectHandle> handle = effect->controlHandle(); 8284 if (handle != 0) { 8285 effect->setEnabled(handle->enabled()); 8286 } 8287 } 8288 desc->mEffect.clear(); 8289 } 8290 mSuspendedEffects.removeItemsAt(index); 8291 } 8292 } 8293} 8294 8295// must be called with ThreadBase::mLock held 8296void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8297{ 8298 sp<SuspendedEffectDesc> desc; 8299 8300 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8301 if (suspend) { 8302 if (index >= 0) { 8303 desc = mSuspendedEffects.valueAt(index); 8304 } else { 8305 desc = new SuspendedEffectDesc(); 8306 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8307 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8308 } 8309 if (desc->mRefCount++ == 0) { 8310 Vector< sp<EffectModule> > effects; 8311 getSuspendEligibleEffects(effects); 8312 for (size_t i = 0; i < effects.size(); i++) { 8313 setEffectSuspended_l(&effects[i]->desc().type, true); 8314 } 8315 } 8316 } else { 8317 if (index < 0) { 8318 return; 8319 } 8320 desc = mSuspendedEffects.valueAt(index); 8321 if (desc->mRefCount <= 0) { 8322 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8323 desc->mRefCount = 1; 8324 } 8325 if (--desc->mRefCount == 0) { 8326 Vector<const effect_uuid_t *> types; 8327 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8328 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8329 continue; 8330 } 8331 types.add(&mSuspendedEffects.valueAt(i)->mType); 8332 } 8333 for (size_t i = 0; i < types.size(); i++) { 8334 setEffectSuspended_l(types[i], false); 8335 } 8336 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8337 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8338 } 8339 } 8340} 8341 8342 8343// The volume effect is used for automated tests only 8344#ifndef OPENSL_ES_H_ 8345static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8346 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8347const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8348#endif //OPENSL_ES_H_ 8349 8350bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8351{ 8352 // auxiliary effects and visualizer are never suspended on output mix 8353 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8354 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8355 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8356 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8357 return false; 8358 } 8359 return true; 8360} 8361 8362void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8363{ 8364 effects.clear(); 8365 for (size_t i = 0; i < mEffects.size(); i++) { 8366 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8367 effects.add(mEffects[i]); 8368 } 8369 } 8370} 8371 8372sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8373 const effect_uuid_t *type) 8374{ 8375 sp<EffectModule> effect = getEffectFromType_l(type); 8376 return effect != 0 && effect->isEnabled() ? effect : 0; 8377} 8378 8379void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8380 bool enabled) 8381{ 8382 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8383 if (enabled) { 8384 if (index < 0) { 8385 // if the effect is not suspend check if all effects are suspended 8386 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8387 if (index < 0) { 8388 return; 8389 } 8390 if (!isEffectEligibleForSuspend(effect->desc())) { 8391 return; 8392 } 8393 setEffectSuspended_l(&effect->desc().type, enabled); 8394 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8395 if (index < 0) { 8396 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8397 return; 8398 } 8399 } 8400 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8401 effect->desc().type.timeLow); 8402 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8403 // if effect is requested to suspended but was not yet enabled, supend it now. 8404 if (desc->mEffect == 0) { 8405 desc->mEffect = effect; 8406 effect->setEnabled(false); 8407 effect->setSuspended(true); 8408 } 8409 } else { 8410 if (index < 0) { 8411 return; 8412 } 8413 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8414 effect->desc().type.timeLow); 8415 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8416 desc->mEffect.clear(); 8417 effect->setSuspended(false); 8418 } 8419} 8420 8421#undef LOG_TAG 8422#define LOG_TAG "AudioFlinger" 8423 8424// ---------------------------------------------------------------------------- 8425 8426status_t AudioFlinger::onTransact( 8427 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8428{ 8429 return BnAudioFlinger::onTransact(code, data, reply, flags); 8430} 8431 8432}; // namespace android 8433