AudioFlinger.cpp revision d3030da2ac3c0ebb8b7bdf38418263caf405b863
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#undef ADD_BATTERY_DATA 41 42#ifdef ADD_BATTERY_DATA 43#include <media/IMediaPlayerService.h> 44#include <media/IMediaDeathNotifier.h> 45#endif 46 47#include <private/media/AudioTrackShared.h> 48#include <private/media/AudioEffectShared.h> 49 50#include <system/audio.h> 51#include <hardware/audio.h> 52 53#include "AudioMixer.h" 54#include "AudioFlinger.h" 55#include "ServiceUtilities.h" 56 57#include <media/EffectsFactoryApi.h> 58#include <audio_effects/effect_visualizer.h> 59#include <audio_effects/effect_ns.h> 60#include <audio_effects/effect_aec.h> 61 62#include <audio_utils/primitives.h> 63 64#include <powermanager/PowerManager.h> 65 66// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72#include <common_time/cc_helper.h> 73#include <common_time/local_clock.h> 74 75// ---------------------------------------------------------------------------- 76 77// Note: the following macro is used for extremely verbose logging message. In 78// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 79// 0; but one side effect of this is to turn all LOGV's as well. Some messages 80// are so verbose that we want to suppress them even when we have ALOG_ASSERT 81// turned on. Do not uncomment the #def below unless you really know what you 82// are doing and want to see all of the extremely verbose messages. 83//#define VERY_VERY_VERBOSE_LOGGING 84#ifdef VERY_VERY_VERBOSE_LOGGING 85#define ALOGVV ALOGV 86#else 87#define ALOGVV(a...) do { } while(0) 88#endif 89 90namespace android { 91 92static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 93static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 94 95static const float MAX_GAIN = 4096.0f; 96static const uint32_t MAX_GAIN_INT = 0x1000; 97 98// retry counts for buffer fill timeout 99// 50 * ~20msecs = 1 second 100static const int8_t kMaxTrackRetries = 50; 101static const int8_t kMaxTrackStartupRetries = 50; 102// allow less retry attempts on direct output thread. 103// direct outputs can be a scarce resource in audio hardware and should 104// be released as quickly as possible. 105static const int8_t kMaxTrackRetriesDirect = 2; 106 107static const int kDumpLockRetries = 50; 108static const int kDumpLockSleepUs = 20000; 109 110// don't warn about blocked writes or record buffer overflows more often than this 111static const nsecs_t kWarningThrottleNs = seconds(5); 112 113// RecordThread loop sleep time upon application overrun or audio HAL read error 114static const int kRecordThreadSleepUs = 5000; 115 116// maximum time to wait for setParameters to complete 117static const nsecs_t kSetParametersTimeoutNs = seconds(2); 118 119// minimum sleep time for the mixer thread loop when tracks are active but in underrun 120static const uint32_t kMinThreadSleepTimeUs = 5000; 121// maximum divider applied to the active sleep time in the mixer thread loop 122static const uint32_t kMaxThreadSleepTimeShift = 2; 123 124nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 125 126// ---------------------------------------------------------------------------- 127 128#ifdef ADD_BATTERY_DATA 129// To collect the amplifier usage 130static void addBatteryData(uint32_t params) { 131 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 132 if (service == NULL) { 133 // it already logged 134 return; 135 } 136 137 service->addBatteryData(params); 138} 139#endif 140 141static int load_audio_interface(const char *if_name, const hw_module_t **mod, 142 audio_hw_device_t **dev) 143{ 144 int rc; 145 146 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 147 if (rc) 148 goto out; 149 150 rc = audio_hw_device_open(*mod, dev); 151 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 152 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 153 if (rc) 154 goto out; 155 156 return 0; 157 158out: 159 *mod = NULL; 160 *dev = NULL; 161 return rc; 162} 163 164// ---------------------------------------------------------------------------- 165 166AudioFlinger::AudioFlinger() 167 : BnAudioFlinger(), 168 mPrimaryHardwareDev(NULL), 169 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 170 mMasterVolume(1.0f), 171 mMasterVolumeSupportLvl(MVS_NONE), 172 mMasterMute(false), 173 mNextUniqueId(1), 174 mMode(AUDIO_MODE_INVALID), 175 mBtNrecIsOff(false) 176{ 177} 178 179void AudioFlinger::onFirstRef() 180{ 181 int rc = 0; 182 183 Mutex::Autolock _l(mLock); 184 185 /* TODO: move all this work into an Init() function */ 186 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 187 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 188 uint32_t int_val; 189 if (1 == sscanf(val_str, "%u", &int_val)) { 190 mStandbyTimeInNsecs = milliseconds(int_val); 191 ALOGI("Using %u mSec as standby time.", int_val); 192 } else { 193 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 194 ALOGI("Using default %u mSec as standby time.", 195 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 196 } 197 } 198 199 mMode = AUDIO_MODE_NORMAL; 200 mMasterVolumeSW = 1.0; 201 mMasterVolume = 1.0; 202 mHardwareStatus = AUDIO_HW_IDLE; 203} 204 205AudioFlinger::~AudioFlinger() 206{ 207 208 while (!mRecordThreads.isEmpty()) { 209 // closeInput() will remove first entry from mRecordThreads 210 closeInput(mRecordThreads.keyAt(0)); 211 } 212 while (!mPlaybackThreads.isEmpty()) { 213 // closeOutput() will remove first entry from mPlaybackThreads 214 closeOutput(mPlaybackThreads.keyAt(0)); 215 } 216 217 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 218 // no mHardwareLock needed, as there are no other references to this 219 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 220 delete mAudioHwDevs.valueAt(i); 221 } 222} 223 224static const char * const audio_interfaces[] = { 225 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 226 AUDIO_HARDWARE_MODULE_ID_A2DP, 227 AUDIO_HARDWARE_MODULE_ID_USB, 228}; 229#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 230 231audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 232{ 233 // if module is 0, the request comes from an old policy manager and we should load 234 // well known modules 235 if (module == 0) { 236 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 237 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 238 loadHwModule_l(audio_interfaces[i]); 239 } 240 } else { 241 // check a match for the requested module handle 242 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 243 if (audioHwdevice != NULL) { 244 return audioHwdevice->hwDevice(); 245 } 246 } 247 // then try to find a module supporting the requested device. 248 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 249 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 250 if ((dev->get_supported_devices(dev) & devices) == devices) 251 return dev; 252 } 253 254 return NULL; 255} 256 257status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 258{ 259 const size_t SIZE = 256; 260 char buffer[SIZE]; 261 String8 result; 262 263 result.append("Clients:\n"); 264 for (size_t i = 0; i < mClients.size(); ++i) { 265 sp<Client> client = mClients.valueAt(i).promote(); 266 if (client != 0) { 267 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 268 result.append(buffer); 269 } 270 } 271 272 result.append("Global session refs:\n"); 273 result.append(" session pid count\n"); 274 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 275 AudioSessionRef *r = mAudioSessionRefs[i]; 276 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 277 result.append(buffer); 278 } 279 write(fd, result.string(), result.size()); 280 return NO_ERROR; 281} 282 283 284status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 285{ 286 const size_t SIZE = 256; 287 char buffer[SIZE]; 288 String8 result; 289 hardware_call_state hardwareStatus = mHardwareStatus; 290 291 snprintf(buffer, SIZE, "Hardware status: %d\n" 292 "Standby Time mSec: %u\n", 293 hardwareStatus, 294 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 295 result.append(buffer); 296 write(fd, result.string(), result.size()); 297 return NO_ERROR; 298} 299 300status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 301{ 302 const size_t SIZE = 256; 303 char buffer[SIZE]; 304 String8 result; 305 snprintf(buffer, SIZE, "Permission Denial: " 306 "can't dump AudioFlinger from pid=%d, uid=%d\n", 307 IPCThreadState::self()->getCallingPid(), 308 IPCThreadState::self()->getCallingUid()); 309 result.append(buffer); 310 write(fd, result.string(), result.size()); 311 return NO_ERROR; 312} 313 314static bool tryLock(Mutex& mutex) 315{ 316 bool locked = false; 317 for (int i = 0; i < kDumpLockRetries; ++i) { 318 if (mutex.tryLock() == NO_ERROR) { 319 locked = true; 320 break; 321 } 322 usleep(kDumpLockSleepUs); 323 } 324 return locked; 325} 326 327status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 328{ 329 if (!dumpAllowed()) { 330 dumpPermissionDenial(fd, args); 331 } else { 332 // get state of hardware lock 333 bool hardwareLocked = tryLock(mHardwareLock); 334 if (!hardwareLocked) { 335 String8 result(kHardwareLockedString); 336 write(fd, result.string(), result.size()); 337 } else { 338 mHardwareLock.unlock(); 339 } 340 341 bool locked = tryLock(mLock); 342 343 // failed to lock - AudioFlinger is probably deadlocked 344 if (!locked) { 345 String8 result(kDeadlockedString); 346 write(fd, result.string(), result.size()); 347 } 348 349 dumpClients(fd, args); 350 dumpInternals(fd, args); 351 352 // dump playback threads 353 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 354 mPlaybackThreads.valueAt(i)->dump(fd, args); 355 } 356 357 // dump record threads 358 for (size_t i = 0; i < mRecordThreads.size(); i++) { 359 mRecordThreads.valueAt(i)->dump(fd, args); 360 } 361 362 // dump all hardware devs 363 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 364 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 365 dev->dump(dev, fd); 366 } 367 if (locked) mLock.unlock(); 368 } 369 return NO_ERROR; 370} 371 372sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 373{ 374 // If pid is already in the mClients wp<> map, then use that entry 375 // (for which promote() is always != 0), otherwise create a new entry and Client. 376 sp<Client> client = mClients.valueFor(pid).promote(); 377 if (client == 0) { 378 client = new Client(this, pid); 379 mClients.add(pid, client); 380 } 381 382 return client; 383} 384 385// IAudioFlinger interface 386 387 388sp<IAudioTrack> AudioFlinger::createTrack( 389 pid_t pid, 390 audio_stream_type_t streamType, 391 uint32_t sampleRate, 392 audio_format_t format, 393 uint32_t channelMask, 394 int frameCount, 395 IAudioFlinger::track_flags_t flags, 396 const sp<IMemory>& sharedBuffer, 397 audio_io_handle_t output, 398 int *sessionId, 399 status_t *status) 400{ 401 sp<PlaybackThread::Track> track; 402 sp<TrackHandle> trackHandle; 403 sp<Client> client; 404 status_t lStatus; 405 int lSessionId; 406 407 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 408 // but if someone uses binder directly they could bypass that and cause us to crash 409 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 410 ALOGE("createTrack() invalid stream type %d", streamType); 411 lStatus = BAD_VALUE; 412 goto Exit; 413 } 414 415 { 416 Mutex::Autolock _l(mLock); 417 PlaybackThread *thread = checkPlaybackThread_l(output); 418 PlaybackThread *effectThread = NULL; 419 if (thread == NULL) { 420 ALOGE("unknown output thread"); 421 lStatus = BAD_VALUE; 422 goto Exit; 423 } 424 425 client = registerPid_l(pid); 426 427 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 428 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 429 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 430 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 431 if (mPlaybackThreads.keyAt(i) != output) { 432 // prevent same audio session on different output threads 433 uint32_t sessions = t->hasAudioSession(*sessionId); 434 if (sessions & PlaybackThread::TRACK_SESSION) { 435 ALOGE("createTrack() session ID %d already in use", *sessionId); 436 lStatus = BAD_VALUE; 437 goto Exit; 438 } 439 // check if an effect with same session ID is waiting for a track to be created 440 if (sessions & PlaybackThread::EFFECT_SESSION) { 441 effectThread = t.get(); 442 } 443 } 444 } 445 lSessionId = *sessionId; 446 } else { 447 // if no audio session id is provided, create one here 448 lSessionId = nextUniqueId(); 449 if (sessionId != NULL) { 450 *sessionId = lSessionId; 451 } 452 } 453 ALOGV("createTrack() lSessionId: %d", lSessionId); 454 455 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 456 track = thread->createTrack_l(client, streamType, sampleRate, format, 457 channelMask, frameCount, sharedBuffer, lSessionId, flags, &lStatus); 458 459 // move effect chain to this output thread if an effect on same session was waiting 460 // for a track to be created 461 if (lStatus == NO_ERROR && effectThread != NULL) { 462 Mutex::Autolock _dl(thread->mLock); 463 Mutex::Autolock _sl(effectThread->mLock); 464 moveEffectChain_l(lSessionId, effectThread, thread, true); 465 } 466 467 // Look for sync events awaiting for a session to be used. 468 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 469 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 470 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 471 track->setSyncEvent(mPendingSyncEvents[i]); 472 mPendingSyncEvents.removeAt(i); 473 i--; 474 } 475 } 476 } 477 } 478 if (lStatus == NO_ERROR) { 479 trackHandle = new TrackHandle(track); 480 } else { 481 // remove local strong reference to Client before deleting the Track so that the Client 482 // destructor is called by the TrackBase destructor with mLock held 483 client.clear(); 484 track.clear(); 485 } 486 487Exit: 488 if (status != NULL) { 489 *status = lStatus; 490 } 491 return trackHandle; 492} 493 494uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 495{ 496 Mutex::Autolock _l(mLock); 497 PlaybackThread *thread = checkPlaybackThread_l(output); 498 if (thread == NULL) { 499 ALOGW("sampleRate() unknown thread %d", output); 500 return 0; 501 } 502 return thread->sampleRate(); 503} 504 505int AudioFlinger::channelCount(audio_io_handle_t output) const 506{ 507 Mutex::Autolock _l(mLock); 508 PlaybackThread *thread = checkPlaybackThread_l(output); 509 if (thread == NULL) { 510 ALOGW("channelCount() unknown thread %d", output); 511 return 0; 512 } 513 return thread->channelCount(); 514} 515 516audio_format_t AudioFlinger::format(audio_io_handle_t output) const 517{ 518 Mutex::Autolock _l(mLock); 519 PlaybackThread *thread = checkPlaybackThread_l(output); 520 if (thread == NULL) { 521 ALOGW("format() unknown thread %d", output); 522 return AUDIO_FORMAT_INVALID; 523 } 524 return thread->format(); 525} 526 527size_t AudioFlinger::frameCount(audio_io_handle_t output) const 528{ 529 Mutex::Autolock _l(mLock); 530 PlaybackThread *thread = checkPlaybackThread_l(output); 531 if (thread == NULL) { 532 ALOGW("frameCount() unknown thread %d", output); 533 return 0; 534 } 535 return thread->frameCount(); 536} 537 538uint32_t AudioFlinger::latency(audio_io_handle_t output) const 539{ 540 Mutex::Autolock _l(mLock); 541 PlaybackThread *thread = checkPlaybackThread_l(output); 542 if (thread == NULL) { 543 ALOGW("latency() unknown thread %d", output); 544 return 0; 545 } 546 return thread->latency(); 547} 548 549status_t AudioFlinger::setMasterVolume(float value) 550{ 551 status_t ret = initCheck(); 552 if (ret != NO_ERROR) { 553 return ret; 554 } 555 556 // check calling permissions 557 if (!settingsAllowed()) { 558 return PERMISSION_DENIED; 559 } 560 561 float swmv = value; 562 563 Mutex::Autolock _l(mLock); 564 565 // when hw supports master volume, don't scale in sw mixer 566 if (MVS_NONE != mMasterVolumeSupportLvl) { 567 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 568 AutoMutex lock(mHardwareLock); 569 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 570 571 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 572 if (NULL != dev->set_master_volume) { 573 dev->set_master_volume(dev, value); 574 } 575 mHardwareStatus = AUDIO_HW_IDLE; 576 } 577 578 swmv = 1.0; 579 } 580 581 mMasterVolume = value; 582 mMasterVolumeSW = swmv; 583 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 584 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 585 586 return NO_ERROR; 587} 588 589status_t AudioFlinger::setMode(audio_mode_t mode) 590{ 591 status_t ret = initCheck(); 592 if (ret != NO_ERROR) { 593 return ret; 594 } 595 596 // check calling permissions 597 if (!settingsAllowed()) { 598 return PERMISSION_DENIED; 599 } 600 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 601 ALOGW("Illegal value: setMode(%d)", mode); 602 return BAD_VALUE; 603 } 604 605 { // scope for the lock 606 AutoMutex lock(mHardwareLock); 607 mHardwareStatus = AUDIO_HW_SET_MODE; 608 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 609 mHardwareStatus = AUDIO_HW_IDLE; 610 } 611 612 if (NO_ERROR == ret) { 613 Mutex::Autolock _l(mLock); 614 mMode = mode; 615 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 616 mPlaybackThreads.valueAt(i)->setMode(mode); 617 } 618 619 return ret; 620} 621 622status_t AudioFlinger::setMicMute(bool state) 623{ 624 status_t ret = initCheck(); 625 if (ret != NO_ERROR) { 626 return ret; 627 } 628 629 // check calling permissions 630 if (!settingsAllowed()) { 631 return PERMISSION_DENIED; 632 } 633 634 AutoMutex lock(mHardwareLock); 635 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 636 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 637 mHardwareStatus = AUDIO_HW_IDLE; 638 return ret; 639} 640 641bool AudioFlinger::getMicMute() const 642{ 643 status_t ret = initCheck(); 644 if (ret != NO_ERROR) { 645 return false; 646 } 647 648 bool state = AUDIO_MODE_INVALID; 649 AutoMutex lock(mHardwareLock); 650 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 651 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 652 mHardwareStatus = AUDIO_HW_IDLE; 653 return state; 654} 655 656status_t AudioFlinger::setMasterMute(bool muted) 657{ 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 Mutex::Autolock _l(mLock); 664 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 665 mMasterMute = muted; 666 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 667 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 668 669 return NO_ERROR; 670} 671 672float AudioFlinger::masterVolume() const 673{ 674 Mutex::Autolock _l(mLock); 675 return masterVolume_l(); 676} 677 678float AudioFlinger::masterVolumeSW() const 679{ 680 Mutex::Autolock _l(mLock); 681 return masterVolumeSW_l(); 682} 683 684bool AudioFlinger::masterMute() const 685{ 686 Mutex::Autolock _l(mLock); 687 return masterMute_l(); 688} 689 690float AudioFlinger::masterVolume_l() const 691{ 692 if (MVS_FULL == mMasterVolumeSupportLvl) { 693 float ret_val; 694 AutoMutex lock(mHardwareLock); 695 696 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 697 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 698 (NULL != mPrimaryHardwareDev->get_master_volume), 699 "can't get master volume"); 700 701 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 702 mHardwareStatus = AUDIO_HW_IDLE; 703 return ret_val; 704 } 705 706 return mMasterVolume; 707} 708 709status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 710 audio_io_handle_t output) 711{ 712 // check calling permissions 713 if (!settingsAllowed()) { 714 return PERMISSION_DENIED; 715 } 716 717 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 718 ALOGE("setStreamVolume() invalid stream %d", stream); 719 return BAD_VALUE; 720 } 721 722 AutoMutex lock(mLock); 723 PlaybackThread *thread = NULL; 724 if (output) { 725 thread = checkPlaybackThread_l(output); 726 if (thread == NULL) { 727 return BAD_VALUE; 728 } 729 } 730 731 mStreamTypes[stream].volume = value; 732 733 if (thread == NULL) { 734 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 735 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 736 } 737 } else { 738 thread->setStreamVolume(stream, value); 739 } 740 741 return NO_ERROR; 742} 743 744status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 745{ 746 // check calling permissions 747 if (!settingsAllowed()) { 748 return PERMISSION_DENIED; 749 } 750 751 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 752 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 753 ALOGE("setStreamMute() invalid stream %d", stream); 754 return BAD_VALUE; 755 } 756 757 AutoMutex lock(mLock); 758 mStreamTypes[stream].mute = muted; 759 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 760 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 761 762 return NO_ERROR; 763} 764 765float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 766{ 767 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 768 return 0.0f; 769 } 770 771 AutoMutex lock(mLock); 772 float volume; 773 if (output) { 774 PlaybackThread *thread = checkPlaybackThread_l(output); 775 if (thread == NULL) { 776 return 0.0f; 777 } 778 volume = thread->streamVolume(stream); 779 } else { 780 volume = streamVolume_l(stream); 781 } 782 783 return volume; 784} 785 786bool AudioFlinger::streamMute(audio_stream_type_t stream) const 787{ 788 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 789 return true; 790 } 791 792 AutoMutex lock(mLock); 793 return streamMute_l(stream); 794} 795 796status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 797{ 798 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 799 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 800 // check calling permissions 801 if (!settingsAllowed()) { 802 return PERMISSION_DENIED; 803 } 804 805 // ioHandle == 0 means the parameters are global to the audio hardware interface 806 if (ioHandle == 0) { 807 Mutex::Autolock _l(mLock); 808 status_t final_result = NO_ERROR; 809 { 810 AutoMutex lock(mHardwareLock); 811 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 812 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 813 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 814 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 815 final_result = result ?: final_result; 816 } 817 mHardwareStatus = AUDIO_HW_IDLE; 818 } 819 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 820 AudioParameter param = AudioParameter(keyValuePairs); 821 String8 value; 822 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 823 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 824 if (mBtNrecIsOff != btNrecIsOff) { 825 for (size_t i = 0; i < mRecordThreads.size(); i++) { 826 sp<RecordThread> thread = mRecordThreads.valueAt(i); 827 RecordThread::RecordTrack *track = thread->track(); 828 if (track != NULL) { 829 audio_devices_t device = (audio_devices_t)( 830 thread->device() & AUDIO_DEVICE_IN_ALL); 831 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 832 thread->setEffectSuspended(FX_IID_AEC, 833 suspend, 834 track->sessionId()); 835 thread->setEffectSuspended(FX_IID_NS, 836 suspend, 837 track->sessionId()); 838 } 839 } 840 mBtNrecIsOff = btNrecIsOff; 841 } 842 } 843 return final_result; 844 } 845 846 // hold a strong ref on thread in case closeOutput() or closeInput() is called 847 // and the thread is exited once the lock is released 848 sp<ThreadBase> thread; 849 { 850 Mutex::Autolock _l(mLock); 851 thread = checkPlaybackThread_l(ioHandle); 852 if (thread == NULL) { 853 thread = checkRecordThread_l(ioHandle); 854 } else if (thread == primaryPlaybackThread_l()) { 855 // indicate output device change to all input threads for pre processing 856 AudioParameter param = AudioParameter(keyValuePairs); 857 int value; 858 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 859 (value != 0)) { 860 for (size_t i = 0; i < mRecordThreads.size(); i++) { 861 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 862 } 863 } 864 } 865 } 866 if (thread != 0) { 867 return thread->setParameters(keyValuePairs); 868 } 869 return BAD_VALUE; 870} 871 872String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 873{ 874// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 875// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 876 877 Mutex::Autolock _l(mLock); 878 879 if (ioHandle == 0) { 880 String8 out_s8; 881 882 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 883 char *s; 884 { 885 AutoMutex lock(mHardwareLock); 886 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 887 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 888 s = dev->get_parameters(dev, keys.string()); 889 mHardwareStatus = AUDIO_HW_IDLE; 890 } 891 out_s8 += String8(s ? s : ""); 892 free(s); 893 } 894 return out_s8; 895 } 896 897 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 898 if (playbackThread != NULL) { 899 return playbackThread->getParameters(keys); 900 } 901 RecordThread *recordThread = checkRecordThread_l(ioHandle); 902 if (recordThread != NULL) { 903 return recordThread->getParameters(keys); 904 } 905 return String8(""); 906} 907 908size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 909{ 910 status_t ret = initCheck(); 911 if (ret != NO_ERROR) { 912 return 0; 913 } 914 915 AutoMutex lock(mHardwareLock); 916 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 917 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 918 mHardwareStatus = AUDIO_HW_IDLE; 919 return size; 920} 921 922unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 923{ 924 if (ioHandle == 0) { 925 return 0; 926 } 927 928 Mutex::Autolock _l(mLock); 929 930 RecordThread *recordThread = checkRecordThread_l(ioHandle); 931 if (recordThread != NULL) { 932 return recordThread->getInputFramesLost(); 933 } 934 return 0; 935} 936 937status_t AudioFlinger::setVoiceVolume(float value) 938{ 939 status_t ret = initCheck(); 940 if (ret != NO_ERROR) { 941 return ret; 942 } 943 944 // check calling permissions 945 if (!settingsAllowed()) { 946 return PERMISSION_DENIED; 947 } 948 949 AutoMutex lock(mHardwareLock); 950 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 951 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 952 mHardwareStatus = AUDIO_HW_IDLE; 953 954 return ret; 955} 956 957status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 958 audio_io_handle_t output) const 959{ 960 status_t status; 961 962 Mutex::Autolock _l(mLock); 963 964 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 965 if (playbackThread != NULL) { 966 return playbackThread->getRenderPosition(halFrames, dspFrames); 967 } 968 969 return BAD_VALUE; 970} 971 972void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 973{ 974 975 Mutex::Autolock _l(mLock); 976 977 pid_t pid = IPCThreadState::self()->getCallingPid(); 978 if (mNotificationClients.indexOfKey(pid) < 0) { 979 sp<NotificationClient> notificationClient = new NotificationClient(this, 980 client, 981 pid); 982 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 983 984 mNotificationClients.add(pid, notificationClient); 985 986 sp<IBinder> binder = client->asBinder(); 987 binder->linkToDeath(notificationClient); 988 989 // the config change is always sent from playback or record threads to avoid deadlock 990 // with AudioSystem::gLock 991 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 992 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 993 } 994 995 for (size_t i = 0; i < mRecordThreads.size(); i++) { 996 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 997 } 998 } 999} 1000 1001void AudioFlinger::removeNotificationClient(pid_t pid) 1002{ 1003 Mutex::Autolock _l(mLock); 1004 1005 mNotificationClients.removeItem(pid); 1006 1007 ALOGV("%d died, releasing its sessions", pid); 1008 size_t num = mAudioSessionRefs.size(); 1009 bool removed = false; 1010 for (size_t i = 0; i< num; ) { 1011 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1012 ALOGV(" pid %d @ %d", ref->mPid, i); 1013 if (ref->mPid == pid) { 1014 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1015 mAudioSessionRefs.removeAt(i); 1016 delete ref; 1017 removed = true; 1018 num--; 1019 } else { 1020 i++; 1021 } 1022 } 1023 if (removed) { 1024 purgeStaleEffects_l(); 1025 } 1026} 1027 1028// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1029void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1030{ 1031 size_t size = mNotificationClients.size(); 1032 for (size_t i = 0; i < size; i++) { 1033 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1034 param2); 1035 } 1036} 1037 1038// removeClient_l() must be called with AudioFlinger::mLock held 1039void AudioFlinger::removeClient_l(pid_t pid) 1040{ 1041 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1042 mClients.removeItem(pid); 1043} 1044 1045 1046// ---------------------------------------------------------------------------- 1047 1048AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1049 uint32_t device, type_t type) 1050 : Thread(false), 1051 mType(type), 1052 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1053 // mChannelMask 1054 mChannelCount(0), 1055 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1056 mParamStatus(NO_ERROR), 1057 mStandby(false), mId(id), 1058 mDevice(device), 1059 mDeathRecipient(new PMDeathRecipient(this)) 1060{ 1061} 1062 1063AudioFlinger::ThreadBase::~ThreadBase() 1064{ 1065 mParamCond.broadcast(); 1066 // do not lock the mutex in destructor 1067 releaseWakeLock_l(); 1068 if (mPowerManager != 0) { 1069 sp<IBinder> binder = mPowerManager->asBinder(); 1070 binder->unlinkToDeath(mDeathRecipient); 1071 } 1072} 1073 1074void AudioFlinger::ThreadBase::exit() 1075{ 1076 ALOGV("ThreadBase::exit"); 1077 { 1078 // This lock prevents the following race in thread (uniprocessor for illustration): 1079 // if (!exitPending()) { 1080 // // context switch from here to exit() 1081 // // exit() calls requestExit(), what exitPending() observes 1082 // // exit() calls signal(), which is dropped since no waiters 1083 // // context switch back from exit() to here 1084 // mWaitWorkCV.wait(...); 1085 // // now thread is hung 1086 // } 1087 AutoMutex lock(mLock); 1088 requestExit(); 1089 mWaitWorkCV.signal(); 1090 } 1091 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1092 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1093 requestExitAndWait(); 1094} 1095 1096status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1097{ 1098 status_t status; 1099 1100 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1101 Mutex::Autolock _l(mLock); 1102 1103 mNewParameters.add(keyValuePairs); 1104 mWaitWorkCV.signal(); 1105 // wait condition with timeout in case the thread loop has exited 1106 // before the request could be processed 1107 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1108 status = mParamStatus; 1109 mWaitWorkCV.signal(); 1110 } else { 1111 status = TIMED_OUT; 1112 } 1113 return status; 1114} 1115 1116void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1117{ 1118 Mutex::Autolock _l(mLock); 1119 sendConfigEvent_l(event, param); 1120} 1121 1122// sendConfigEvent_l() must be called with ThreadBase::mLock held 1123void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1124{ 1125 ConfigEvent configEvent; 1126 configEvent.mEvent = event; 1127 configEvent.mParam = param; 1128 mConfigEvents.add(configEvent); 1129 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1130 mWaitWorkCV.signal(); 1131} 1132 1133void AudioFlinger::ThreadBase::processConfigEvents() 1134{ 1135 mLock.lock(); 1136 while (!mConfigEvents.isEmpty()) { 1137 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1138 ConfigEvent configEvent = mConfigEvents[0]; 1139 mConfigEvents.removeAt(0); 1140 // release mLock before locking AudioFlinger mLock: lock order is always 1141 // AudioFlinger then ThreadBase to avoid cross deadlock 1142 mLock.unlock(); 1143 mAudioFlinger->mLock.lock(); 1144 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1145 mAudioFlinger->mLock.unlock(); 1146 mLock.lock(); 1147 } 1148 mLock.unlock(); 1149} 1150 1151status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1152{ 1153 const size_t SIZE = 256; 1154 char buffer[SIZE]; 1155 String8 result; 1156 1157 bool locked = tryLock(mLock); 1158 if (!locked) { 1159 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1160 write(fd, buffer, strlen(buffer)); 1161 } 1162 1163 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1164 result.append(buffer); 1165 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1166 result.append(buffer); 1167 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1168 result.append(buffer); 1169 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1170 result.append(buffer); 1171 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1172 result.append(buffer); 1173 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1174 result.append(buffer); 1175 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1176 result.append(buffer); 1177 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1178 result.append(buffer); 1179 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1180 result.append(buffer); 1181 1182 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1183 result.append(buffer); 1184 result.append(" Index Command"); 1185 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1186 snprintf(buffer, SIZE, "\n %02d ", i); 1187 result.append(buffer); 1188 result.append(mNewParameters[i]); 1189 } 1190 1191 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1192 result.append(buffer); 1193 snprintf(buffer, SIZE, " Index event param\n"); 1194 result.append(buffer); 1195 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1196 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1197 result.append(buffer); 1198 } 1199 result.append("\n"); 1200 1201 write(fd, result.string(), result.size()); 1202 1203 if (locked) { 1204 mLock.unlock(); 1205 } 1206 return NO_ERROR; 1207} 1208 1209status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1210{ 1211 const size_t SIZE = 256; 1212 char buffer[SIZE]; 1213 String8 result; 1214 1215 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1216 write(fd, buffer, strlen(buffer)); 1217 1218 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1219 sp<EffectChain> chain = mEffectChains[i]; 1220 if (chain != 0) { 1221 chain->dump(fd, args); 1222 } 1223 } 1224 return NO_ERROR; 1225} 1226 1227void AudioFlinger::ThreadBase::acquireWakeLock() 1228{ 1229 Mutex::Autolock _l(mLock); 1230 acquireWakeLock_l(); 1231} 1232 1233void AudioFlinger::ThreadBase::acquireWakeLock_l() 1234{ 1235 if (mPowerManager == 0) { 1236 // use checkService() to avoid blocking if power service is not up yet 1237 sp<IBinder> binder = 1238 defaultServiceManager()->checkService(String16("power")); 1239 if (binder == 0) { 1240 ALOGW("Thread %s cannot connect to the power manager service", mName); 1241 } else { 1242 mPowerManager = interface_cast<IPowerManager>(binder); 1243 binder->linkToDeath(mDeathRecipient); 1244 } 1245 } 1246 if (mPowerManager != 0) { 1247 sp<IBinder> binder = new BBinder(); 1248 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1249 binder, 1250 String16(mName)); 1251 if (status == NO_ERROR) { 1252 mWakeLockToken = binder; 1253 } 1254 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1255 } 1256} 1257 1258void AudioFlinger::ThreadBase::releaseWakeLock() 1259{ 1260 Mutex::Autolock _l(mLock); 1261 releaseWakeLock_l(); 1262} 1263 1264void AudioFlinger::ThreadBase::releaseWakeLock_l() 1265{ 1266 if (mWakeLockToken != 0) { 1267 ALOGV("releaseWakeLock_l() %s", mName); 1268 if (mPowerManager != 0) { 1269 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1270 } 1271 mWakeLockToken.clear(); 1272 } 1273} 1274 1275void AudioFlinger::ThreadBase::clearPowerManager() 1276{ 1277 Mutex::Autolock _l(mLock); 1278 releaseWakeLock_l(); 1279 mPowerManager.clear(); 1280} 1281 1282void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1283{ 1284 sp<ThreadBase> thread = mThread.promote(); 1285 if (thread != 0) { 1286 thread->clearPowerManager(); 1287 } 1288 ALOGW("power manager service died !!!"); 1289} 1290 1291void AudioFlinger::ThreadBase::setEffectSuspended( 1292 const effect_uuid_t *type, bool suspend, int sessionId) 1293{ 1294 Mutex::Autolock _l(mLock); 1295 setEffectSuspended_l(type, suspend, sessionId); 1296} 1297 1298void AudioFlinger::ThreadBase::setEffectSuspended_l( 1299 const effect_uuid_t *type, bool suspend, int sessionId) 1300{ 1301 sp<EffectChain> chain = getEffectChain_l(sessionId); 1302 if (chain != 0) { 1303 if (type != NULL) { 1304 chain->setEffectSuspended_l(type, suspend); 1305 } else { 1306 chain->setEffectSuspendedAll_l(suspend); 1307 } 1308 } 1309 1310 updateSuspendedSessions_l(type, suspend, sessionId); 1311} 1312 1313void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1314{ 1315 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1316 if (index < 0) { 1317 return; 1318 } 1319 1320 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1321 mSuspendedSessions.editValueAt(index); 1322 1323 for (size_t i = 0; i < sessionEffects.size(); i++) { 1324 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1325 for (int j = 0; j < desc->mRefCount; j++) { 1326 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1327 chain->setEffectSuspendedAll_l(true); 1328 } else { 1329 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1330 desc->mType.timeLow); 1331 chain->setEffectSuspended_l(&desc->mType, true); 1332 } 1333 } 1334 } 1335} 1336 1337void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1338 bool suspend, 1339 int sessionId) 1340{ 1341 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1342 1343 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1344 1345 if (suspend) { 1346 if (index >= 0) { 1347 sessionEffects = mSuspendedSessions.editValueAt(index); 1348 } else { 1349 mSuspendedSessions.add(sessionId, sessionEffects); 1350 } 1351 } else { 1352 if (index < 0) { 1353 return; 1354 } 1355 sessionEffects = mSuspendedSessions.editValueAt(index); 1356 } 1357 1358 1359 int key = EffectChain::kKeyForSuspendAll; 1360 if (type != NULL) { 1361 key = type->timeLow; 1362 } 1363 index = sessionEffects.indexOfKey(key); 1364 1365 sp<SuspendedSessionDesc> desc; 1366 if (suspend) { 1367 if (index >= 0) { 1368 desc = sessionEffects.valueAt(index); 1369 } else { 1370 desc = new SuspendedSessionDesc(); 1371 if (type != NULL) { 1372 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1373 } 1374 sessionEffects.add(key, desc); 1375 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1376 } 1377 desc->mRefCount++; 1378 } else { 1379 if (index < 0) { 1380 return; 1381 } 1382 desc = sessionEffects.valueAt(index); 1383 if (--desc->mRefCount == 0) { 1384 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1385 sessionEffects.removeItemsAt(index); 1386 if (sessionEffects.isEmpty()) { 1387 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1388 sessionId); 1389 mSuspendedSessions.removeItem(sessionId); 1390 } 1391 } 1392 } 1393 if (!sessionEffects.isEmpty()) { 1394 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1395 } 1396} 1397 1398void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1399 bool enabled, 1400 int sessionId) 1401{ 1402 Mutex::Autolock _l(mLock); 1403 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1404} 1405 1406void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1407 bool enabled, 1408 int sessionId) 1409{ 1410 if (mType != RECORD) { 1411 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1412 // another session. This gives the priority to well behaved effect control panels 1413 // and applications not using global effects. 1414 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1415 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1416 } 1417 } 1418 1419 sp<EffectChain> chain = getEffectChain_l(sessionId); 1420 if (chain != 0) { 1421 chain->checkSuspendOnEffectEnabled(effect, enabled); 1422 } 1423} 1424 1425// ---------------------------------------------------------------------------- 1426 1427AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1428 AudioStreamOut* output, 1429 audio_io_handle_t id, 1430 uint32_t device, 1431 type_t type) 1432 : ThreadBase(audioFlinger, id, device, type), 1433 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1434 // Assumes constructor is called by AudioFlinger with it's mLock held, 1435 // but it would be safer to explicitly pass initial masterMute as parameter 1436 mMasterMute(audioFlinger->masterMute_l()), 1437 // mStreamTypes[] initialized in constructor body 1438 mOutput(output), 1439 // Assumes constructor is called by AudioFlinger with it's mLock held, 1440 // but it would be safer to explicitly pass initial masterVolume as parameter 1441 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1442 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1443 mMixerStatus(MIXER_IDLE), 1444 mPrevMixerStatus(MIXER_IDLE), 1445 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1446{ 1447 snprintf(mName, kNameLength, "AudioOut_%X", id); 1448 1449 readOutputParameters(); 1450 1451 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1452 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1453 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1454 stream = (audio_stream_type_t) (stream + 1)) { 1455 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1456 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1457 } 1458 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1459 // because mAudioFlinger doesn't have one to copy from 1460} 1461 1462AudioFlinger::PlaybackThread::~PlaybackThread() 1463{ 1464 delete [] mMixBuffer; 1465} 1466 1467status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1468{ 1469 dumpInternals(fd, args); 1470 dumpTracks(fd, args); 1471 dumpEffectChains(fd, args); 1472 return NO_ERROR; 1473} 1474 1475status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1476{ 1477 const size_t SIZE = 256; 1478 char buffer[SIZE]; 1479 String8 result; 1480 1481 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1482 result.append(buffer); 1483 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1484 for (size_t i = 0; i < mTracks.size(); ++i) { 1485 sp<Track> track = mTracks[i]; 1486 if (track != 0) { 1487 track->dump(buffer, SIZE); 1488 result.append(buffer); 1489 } 1490 } 1491 1492 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1493 result.append(buffer); 1494 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1495 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1496 sp<Track> track = mActiveTracks[i].promote(); 1497 if (track != 0) { 1498 track->dump(buffer, SIZE); 1499 result.append(buffer); 1500 } 1501 } 1502 write(fd, result.string(), result.size()); 1503 return NO_ERROR; 1504} 1505 1506status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1507{ 1508 const size_t SIZE = 256; 1509 char buffer[SIZE]; 1510 String8 result; 1511 1512 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1513 result.append(buffer); 1514 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1515 result.append(buffer); 1516 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1517 result.append(buffer); 1518 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1519 result.append(buffer); 1520 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1521 result.append(buffer); 1522 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1523 result.append(buffer); 1524 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1525 result.append(buffer); 1526 write(fd, result.string(), result.size()); 1527 1528 dumpBase(fd, args); 1529 1530 return NO_ERROR; 1531} 1532 1533// Thread virtuals 1534status_t AudioFlinger::PlaybackThread::readyToRun() 1535{ 1536 status_t status = initCheck(); 1537 if (status == NO_ERROR) { 1538 ALOGI("AudioFlinger's thread %p ready to run", this); 1539 } else { 1540 ALOGE("No working audio driver found."); 1541 } 1542 return status; 1543} 1544 1545void AudioFlinger::PlaybackThread::onFirstRef() 1546{ 1547 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1548} 1549 1550// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1551sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1552 const sp<AudioFlinger::Client>& client, 1553 audio_stream_type_t streamType, 1554 uint32_t sampleRate, 1555 audio_format_t format, 1556 uint32_t channelMask, 1557 int frameCount, 1558 const sp<IMemory>& sharedBuffer, 1559 int sessionId, 1560 IAudioFlinger::track_flags_t flags, 1561 status_t *status) 1562{ 1563 sp<Track> track; 1564 status_t lStatus; 1565 1566 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1567 1568 // client expresses a preference for FAST, but we get the final say 1569 if ((flags & IAudioFlinger::TRACK_FAST) && 1570 !( 1571 // not timed 1572 (!isTimed) && 1573 // either of these use cases: 1574 ( 1575 // use case 1: shared buffer with any frame count 1576 ( 1577 (sharedBuffer != 0) 1578 ) || 1579 // use case 2: callback handler and small power-of-2 frame count 1580 ( 1581 // unfortunately we can't verify that there's a callback until start() 1582 // FIXME supported frame counts should not be hard-coded 1583 ( 1584 (frameCount == 128) || 1585 (frameCount == 256) || 1586 (frameCount == 512) 1587 ) 1588 ) 1589 ) && 1590 // PCM data 1591 audio_is_linear_pcm(format) && 1592 // mono or stereo 1593 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1594 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1595 // hardware sample rate 1596 (sampleRate == mSampleRate) 1597 // FIXME test that MixerThread for this fast track has a capable output HAL 1598 // FIXME add a permission test also? 1599 ) ) { 1600 ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied"); 1601 flags &= ~IAudioFlinger::TRACK_FAST; 1602 } 1603 1604 if (mType == DIRECT) { 1605 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1606 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1607 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1608 "for output %p with format %d", 1609 sampleRate, format, channelMask, mOutput, mFormat); 1610 lStatus = BAD_VALUE; 1611 goto Exit; 1612 } 1613 } 1614 } else { 1615 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1616 if (sampleRate > mSampleRate*2) { 1617 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1618 lStatus = BAD_VALUE; 1619 goto Exit; 1620 } 1621 } 1622 1623 lStatus = initCheck(); 1624 if (lStatus != NO_ERROR) { 1625 ALOGE("Audio driver not initialized."); 1626 goto Exit; 1627 } 1628 1629 { // scope for mLock 1630 Mutex::Autolock _l(mLock); 1631 1632 // all tracks in same audio session must share the same routing strategy otherwise 1633 // conflicts will happen when tracks are moved from one output to another by audio policy 1634 // manager 1635 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1636 for (size_t i = 0; i < mTracks.size(); ++i) { 1637 sp<Track> t = mTracks[i]; 1638 if (t != 0 && !t->isOutputTrack()) { 1639 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1640 if (sessionId == t->sessionId() && strategy != actual) { 1641 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1642 strategy, actual); 1643 lStatus = BAD_VALUE; 1644 goto Exit; 1645 } 1646 } 1647 } 1648 1649 if (!isTimed) { 1650 track = new Track(this, client, streamType, sampleRate, format, 1651 channelMask, frameCount, sharedBuffer, sessionId, flags); 1652 } else { 1653 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1654 channelMask, frameCount, sharedBuffer, sessionId); 1655 } 1656 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1657 lStatus = NO_MEMORY; 1658 goto Exit; 1659 } 1660 mTracks.add(track); 1661 1662 sp<EffectChain> chain = getEffectChain_l(sessionId); 1663 if (chain != 0) { 1664 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1665 track->setMainBuffer(chain->inBuffer()); 1666 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1667 chain->incTrackCnt(); 1668 } 1669 } 1670 lStatus = NO_ERROR; 1671 1672Exit: 1673 if (status) { 1674 *status = lStatus; 1675 } 1676 return track; 1677} 1678 1679uint32_t AudioFlinger::PlaybackThread::latency() const 1680{ 1681 Mutex::Autolock _l(mLock); 1682 if (initCheck() == NO_ERROR) { 1683 return mOutput->stream->get_latency(mOutput->stream); 1684 } else { 1685 return 0; 1686 } 1687} 1688 1689void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1690{ 1691 Mutex::Autolock _l(mLock); 1692 mMasterVolume = value; 1693} 1694 1695void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1696{ 1697 Mutex::Autolock _l(mLock); 1698 setMasterMute_l(muted); 1699} 1700 1701void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1702{ 1703 Mutex::Autolock _l(mLock); 1704 mStreamTypes[stream].volume = value; 1705} 1706 1707void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1708{ 1709 Mutex::Autolock _l(mLock); 1710 mStreamTypes[stream].mute = muted; 1711} 1712 1713float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1714{ 1715 Mutex::Autolock _l(mLock); 1716 return mStreamTypes[stream].volume; 1717} 1718 1719// addTrack_l() must be called with ThreadBase::mLock held 1720status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1721{ 1722 status_t status = ALREADY_EXISTS; 1723 1724 // set retry count for buffer fill 1725 track->mRetryCount = kMaxTrackStartupRetries; 1726 if (mActiveTracks.indexOf(track) < 0) { 1727 // the track is newly added, make sure it fills up all its 1728 // buffers before playing. This is to ensure the client will 1729 // effectively get the latency it requested. 1730 track->mFillingUpStatus = Track::FS_FILLING; 1731 track->mResetDone = false; 1732 mActiveTracks.add(track); 1733 if (track->mainBuffer() != mMixBuffer) { 1734 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1735 if (chain != 0) { 1736 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1737 chain->incActiveTrackCnt(); 1738 } 1739 } 1740 1741 status = NO_ERROR; 1742 } 1743 1744 ALOGV("mWaitWorkCV.broadcast"); 1745 mWaitWorkCV.broadcast(); 1746 1747 return status; 1748} 1749 1750// destroyTrack_l() must be called with ThreadBase::mLock held 1751void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1752{ 1753 track->mState = TrackBase::TERMINATED; 1754 if (mActiveTracks.indexOf(track) < 0) { 1755 removeTrack_l(track); 1756 } 1757} 1758 1759void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1760{ 1761 mTracks.remove(track); 1762 deleteTrackName_l(track->name()); 1763 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1764 if (chain != 0) { 1765 chain->decTrackCnt(); 1766 } 1767} 1768 1769String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1770{ 1771 String8 out_s8 = String8(""); 1772 char *s; 1773 1774 Mutex::Autolock _l(mLock); 1775 if (initCheck() != NO_ERROR) { 1776 return out_s8; 1777 } 1778 1779 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1780 out_s8 = String8(s); 1781 free(s); 1782 return out_s8; 1783} 1784 1785// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1786void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1787 AudioSystem::OutputDescriptor desc; 1788 void *param2 = NULL; 1789 1790 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1791 1792 switch (event) { 1793 case AudioSystem::OUTPUT_OPENED: 1794 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1795 desc.channels = mChannelMask; 1796 desc.samplingRate = mSampleRate; 1797 desc.format = mFormat; 1798 desc.frameCount = mFrameCount; 1799 desc.latency = latency(); 1800 param2 = &desc; 1801 break; 1802 1803 case AudioSystem::STREAM_CONFIG_CHANGED: 1804 param2 = ¶m; 1805 case AudioSystem::OUTPUT_CLOSED: 1806 default: 1807 break; 1808 } 1809 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1810} 1811 1812void AudioFlinger::PlaybackThread::readOutputParameters() 1813{ 1814 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1815 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1816 mChannelCount = (uint16_t)popcount(mChannelMask); 1817 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1818 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1819 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1820 1821 // FIXME - Current mixer implementation only supports stereo output: Always 1822 // Allocate a stereo buffer even if HW output is mono. 1823 delete[] mMixBuffer; 1824 mMixBuffer = new int16_t[mFrameCount * 2]; 1825 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1826 1827 // force reconfiguration of effect chains and engines to take new buffer size and audio 1828 // parameters into account 1829 // Note that mLock is not held when readOutputParameters() is called from the constructor 1830 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1831 // matter. 1832 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1833 Vector< sp<EffectChain> > effectChains = mEffectChains; 1834 for (size_t i = 0; i < effectChains.size(); i ++) { 1835 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1836 } 1837} 1838 1839status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1840{ 1841 if (halFrames == NULL || dspFrames == NULL) { 1842 return BAD_VALUE; 1843 } 1844 Mutex::Autolock _l(mLock); 1845 if (initCheck() != NO_ERROR) { 1846 return INVALID_OPERATION; 1847 } 1848 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1849 1850 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1851} 1852 1853uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1854{ 1855 Mutex::Autolock _l(mLock); 1856 uint32_t result = 0; 1857 if (getEffectChain_l(sessionId) != 0) { 1858 result = EFFECT_SESSION; 1859 } 1860 1861 for (size_t i = 0; i < mTracks.size(); ++i) { 1862 sp<Track> track = mTracks[i]; 1863 if (sessionId == track->sessionId() && 1864 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1865 result |= TRACK_SESSION; 1866 break; 1867 } 1868 } 1869 1870 return result; 1871} 1872 1873uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1874{ 1875 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1876 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1877 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1878 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1879 } 1880 for (size_t i = 0; i < mTracks.size(); i++) { 1881 sp<Track> track = mTracks[i]; 1882 if (sessionId == track->sessionId() && 1883 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1884 return AudioSystem::getStrategyForStream(track->streamType()); 1885 } 1886 } 1887 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1888} 1889 1890 1891AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1892{ 1893 Mutex::Autolock _l(mLock); 1894 return mOutput; 1895} 1896 1897AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1898{ 1899 Mutex::Autolock _l(mLock); 1900 AudioStreamOut *output = mOutput; 1901 mOutput = NULL; 1902 return output; 1903} 1904 1905// this method must always be called either with ThreadBase mLock held or inside the thread loop 1906audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1907{ 1908 if (mOutput == NULL) { 1909 return NULL; 1910 } 1911 return &mOutput->stream->common; 1912} 1913 1914uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1915{ 1916 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1917 // decoding and transfer time. So sleeping for half of the latency would likely cause 1918 // underruns 1919 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1920 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1921 } else { 1922 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1923 } 1924} 1925 1926status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1927{ 1928 if (!isValidSyncEvent(event)) { 1929 return BAD_VALUE; 1930 } 1931 1932 Mutex::Autolock _l(mLock); 1933 1934 for (size_t i = 0; i < mTracks.size(); ++i) { 1935 sp<Track> track = mTracks[i]; 1936 if (event->triggerSession() == track->sessionId()) { 1937 track->setSyncEvent(event); 1938 return NO_ERROR; 1939 } 1940 } 1941 1942 return NAME_NOT_FOUND; 1943} 1944 1945bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 1946{ 1947 switch (event->type()) { 1948 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 1949 return true; 1950 default: 1951 break; 1952 } 1953 return false; 1954} 1955 1956// ---------------------------------------------------------------------------- 1957 1958AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1959 audio_io_handle_t id, uint32_t device, type_t type) 1960 : PlaybackThread(audioFlinger, output, id, device, type) 1961{ 1962 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1963 // FIXME - Current mixer implementation only supports stereo output 1964 if (mChannelCount == 1) { 1965 ALOGE("Invalid audio hardware channel count"); 1966 } 1967} 1968 1969AudioFlinger::MixerThread::~MixerThread() 1970{ 1971 delete mAudioMixer; 1972} 1973 1974class CpuStats { 1975public: 1976 CpuStats(); 1977 void sample(const String8 &title); 1978#ifdef DEBUG_CPU_USAGE 1979private: 1980 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 1981 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 1982 1983 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 1984 1985 int mCpuNum; // thread's current CPU number 1986 int mCpukHz; // frequency of thread's current CPU in kHz 1987#endif 1988}; 1989 1990CpuStats::CpuStats() 1991#ifdef DEBUG_CPU_USAGE 1992 : mCpuNum(-1), mCpukHz(-1) 1993#endif 1994{ 1995} 1996 1997void CpuStats::sample(const String8 &title) { 1998#ifdef DEBUG_CPU_USAGE 1999 // get current thread's delta CPU time in wall clock ns 2000 double wcNs; 2001 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2002 2003 // record sample for wall clock statistics 2004 if (valid) { 2005 mWcStats.sample(wcNs); 2006 } 2007 2008 // get the current CPU number 2009 int cpuNum = sched_getcpu(); 2010 2011 // get the current CPU frequency in kHz 2012 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2013 2014 // check if either CPU number or frequency changed 2015 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2016 mCpuNum = cpuNum; 2017 mCpukHz = cpukHz; 2018 // ignore sample for purposes of cycles 2019 valid = false; 2020 } 2021 2022 // if no change in CPU number or frequency, then record sample for cycle statistics 2023 if (valid && mCpukHz > 0) { 2024 double cycles = wcNs * cpukHz * 0.000001; 2025 mHzStats.sample(cycles); 2026 } 2027 2028 unsigned n = mWcStats.n(); 2029 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2030 if ((n & 127) == 1) { 2031 long long elapsed = mCpuUsage.elapsed(); 2032 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2033 double perLoop = elapsed / (double) n; 2034 double perLoop100 = perLoop * 0.01; 2035 double perLoop1k = perLoop * 0.001; 2036 double mean = mWcStats.mean(); 2037 double stddev = mWcStats.stddev(); 2038 double minimum = mWcStats.minimum(); 2039 double maximum = mWcStats.maximum(); 2040 double meanCycles = mHzStats.mean(); 2041 double stddevCycles = mHzStats.stddev(); 2042 double minCycles = mHzStats.minimum(); 2043 double maxCycles = mHzStats.maximum(); 2044 mCpuUsage.resetElapsed(); 2045 mWcStats.reset(); 2046 mHzStats.reset(); 2047 ALOGD("CPU usage for %s over past %.1f secs\n" 2048 " (%u mixer loops at %.1f mean ms per loop):\n" 2049 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2050 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2051 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2052 title.string(), 2053 elapsed * .000000001, n, perLoop * .000001, 2054 mean * .001, 2055 stddev * .001, 2056 minimum * .001, 2057 maximum * .001, 2058 mean / perLoop100, 2059 stddev / perLoop100, 2060 minimum / perLoop100, 2061 maximum / perLoop100, 2062 meanCycles / perLoop1k, 2063 stddevCycles / perLoop1k, 2064 minCycles / perLoop1k, 2065 maxCycles / perLoop1k); 2066 2067 } 2068 } 2069#endif 2070}; 2071 2072void AudioFlinger::PlaybackThread::checkSilentMode_l() 2073{ 2074 if (!mMasterMute) { 2075 char value[PROPERTY_VALUE_MAX]; 2076 if (property_get("ro.audio.silent", value, "0") > 0) { 2077 char *endptr; 2078 unsigned long ul = strtoul(value, &endptr, 0); 2079 if (*endptr == '\0' && ul != 0) { 2080 ALOGD("Silence is golden"); 2081 // The setprop command will not allow a property to be changed after 2082 // the first time it is set, so we don't have to worry about un-muting. 2083 setMasterMute_l(true); 2084 } 2085 } 2086 } 2087} 2088 2089bool AudioFlinger::PlaybackThread::threadLoop() 2090{ 2091 Vector< sp<Track> > tracksToRemove; 2092 2093 standbyTime = systemTime(); 2094 2095 // MIXER 2096 nsecs_t lastWarning = 0; 2097if (mType == MIXER) { 2098 longStandbyExit = false; 2099} 2100 2101 // DUPLICATING 2102 // FIXME could this be made local to while loop? 2103 writeFrames = 0; 2104 2105 cacheParameters_l(); 2106 sleepTime = idleSleepTime; 2107 2108if (mType == MIXER) { 2109 sleepTimeShift = 0; 2110} 2111 2112 CpuStats cpuStats; 2113 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2114 2115 acquireWakeLock(); 2116 2117 while (!exitPending()) 2118 { 2119 cpuStats.sample(myName); 2120 2121 Vector< sp<EffectChain> > effectChains; 2122 2123 processConfigEvents(); 2124 2125 { // scope for mLock 2126 2127 Mutex::Autolock _l(mLock); 2128 2129 if (checkForNewParameters_l()) { 2130 cacheParameters_l(); 2131 } 2132 2133 saveOutputTracks(); 2134 2135 // put audio hardware into standby after short delay 2136 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2137 mSuspended > 0)) { 2138 if (!mStandby) { 2139 2140 threadLoop_standby(); 2141 2142 mStandby = true; 2143 mBytesWritten = 0; 2144 } 2145 2146 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2147 // we're about to wait, flush the binder command buffer 2148 IPCThreadState::self()->flushCommands(); 2149 2150 clearOutputTracks(); 2151 2152 if (exitPending()) break; 2153 2154 releaseWakeLock_l(); 2155 // wait until we have something to do... 2156 ALOGV("%s going to sleep", myName.string()); 2157 mWaitWorkCV.wait(mLock); 2158 ALOGV("%s waking up", myName.string()); 2159 acquireWakeLock_l(); 2160 2161 mPrevMixerStatus = MIXER_IDLE; 2162 2163 checkSilentMode_l(); 2164 2165 standbyTime = systemTime() + standbyDelay; 2166 sleepTime = idleSleepTime; 2167 if (mType == MIXER) { 2168 sleepTimeShift = 0; 2169 } 2170 2171 continue; 2172 } 2173 } 2174 2175 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2176 // Shift in the new status; this could be a queue if it's 2177 // useful to filter the mixer status over several cycles. 2178 mPrevMixerStatus = mMixerStatus; 2179 mMixerStatus = newMixerStatus; 2180 2181 // prevent any changes in effect chain list and in each effect chain 2182 // during mixing and effect process as the audio buffers could be deleted 2183 // or modified if an effect is created or deleted 2184 lockEffectChains_l(effectChains); 2185 } 2186 2187 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2188 threadLoop_mix(); 2189 } else { 2190 threadLoop_sleepTime(); 2191 } 2192 2193 if (mSuspended > 0) { 2194 sleepTime = suspendSleepTimeUs(); 2195 } 2196 2197 // only process effects if we're going to write 2198 if (sleepTime == 0) { 2199 for (size_t i = 0; i < effectChains.size(); i ++) { 2200 effectChains[i]->process_l(); 2201 } 2202 } 2203 2204 // enable changes in effect chain 2205 unlockEffectChains(effectChains); 2206 2207 // sleepTime == 0 means we must write to audio hardware 2208 if (sleepTime == 0) { 2209 2210 threadLoop_write(); 2211 2212if (mType == MIXER) { 2213 // write blocked detection 2214 nsecs_t now = systemTime(); 2215 nsecs_t delta = now - mLastWriteTime; 2216 if (!mStandby && delta > maxPeriod) { 2217 mNumDelayedWrites++; 2218 if ((now - lastWarning) > kWarningThrottleNs) { 2219 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2220 ns2ms(delta), mNumDelayedWrites, this); 2221 lastWarning = now; 2222 } 2223 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2224 // a different threshold. Or completely removed for what it is worth anyway... 2225 if (mStandby) { 2226 longStandbyExit = true; 2227 } 2228 } 2229} 2230 2231 mStandby = false; 2232 } else { 2233 usleep(sleepTime); 2234 } 2235 2236 // finally let go of removed track(s), without the lock held 2237 // since we can't guarantee the destructors won't acquire that 2238 // same lock. 2239 tracksToRemove.clear(); 2240 2241 // FIXME I don't understand the need for this here; 2242 // it was in the original code but maybe the 2243 // assignment in saveOutputTracks() makes this unnecessary? 2244 clearOutputTracks(); 2245 2246 // Effect chains will be actually deleted here if they were removed from 2247 // mEffectChains list during mixing or effects processing 2248 effectChains.clear(); 2249 2250 // FIXME Note that the above .clear() is no longer necessary since effectChains 2251 // is now local to this block, but will keep it for now (at least until merge done). 2252 } 2253 2254if (mType == MIXER || mType == DIRECT) { 2255 // put output stream into standby mode 2256 if (!mStandby) { 2257 mOutput->stream->common.standby(&mOutput->stream->common); 2258 } 2259} 2260if (mType == DUPLICATING) { 2261 // for DuplicatingThread, standby mode is handled by the outputTracks 2262} 2263 2264 releaseWakeLock(); 2265 2266 ALOGV("Thread %p type %d exiting", this, mType); 2267 return false; 2268} 2269 2270// shared by MIXER and DIRECT, overridden by DUPLICATING 2271void AudioFlinger::PlaybackThread::threadLoop_write() 2272{ 2273 // FIXME rewrite to reduce number of system calls 2274 mLastWriteTime = systemTime(); 2275 mInWrite = true; 2276 mBytesWritten += mixBufferSize; 2277 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2278 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2279 mNumWrites++; 2280 mInWrite = false; 2281} 2282 2283// shared by MIXER and DIRECT, overridden by DUPLICATING 2284void AudioFlinger::PlaybackThread::threadLoop_standby() 2285{ 2286 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2287 mOutput->stream->common.standby(&mOutput->stream->common); 2288} 2289 2290void AudioFlinger::MixerThread::threadLoop_mix() 2291{ 2292 // obtain the presentation timestamp of the next output buffer 2293 int64_t pts; 2294 status_t status = INVALID_OPERATION; 2295 2296 if (NULL != mOutput->stream->get_next_write_timestamp) { 2297 status = mOutput->stream->get_next_write_timestamp( 2298 mOutput->stream, &pts); 2299 } 2300 2301 if (status != NO_ERROR) { 2302 pts = AudioBufferProvider::kInvalidPTS; 2303 } 2304 2305 // mix buffers... 2306 mAudioMixer->process(pts); 2307 // increase sleep time progressively when application underrun condition clears. 2308 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2309 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2310 // such that we would underrun the audio HAL. 2311 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2312 sleepTimeShift--; 2313 } 2314 sleepTime = 0; 2315 standbyTime = systemTime() + standbyDelay; 2316 //TODO: delay standby when effects have a tail 2317} 2318 2319void AudioFlinger::MixerThread::threadLoop_sleepTime() 2320{ 2321 // If no tracks are ready, sleep once for the duration of an output 2322 // buffer size, then write 0s to the output 2323 if (sleepTime == 0) { 2324 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2325 sleepTime = activeSleepTime >> sleepTimeShift; 2326 if (sleepTime < kMinThreadSleepTimeUs) { 2327 sleepTime = kMinThreadSleepTimeUs; 2328 } 2329 // reduce sleep time in case of consecutive application underruns to avoid 2330 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2331 // duration we would end up writing less data than needed by the audio HAL if 2332 // the condition persists. 2333 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2334 sleepTimeShift++; 2335 } 2336 } else { 2337 sleepTime = idleSleepTime; 2338 } 2339 } else if (mBytesWritten != 0 || 2340 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2341 memset (mMixBuffer, 0, mixBufferSize); 2342 sleepTime = 0; 2343 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2344 } 2345 // TODO add standby time extension fct of effect tail 2346} 2347 2348// prepareTracks_l() must be called with ThreadBase::mLock held 2349AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2350 Vector< sp<Track> > *tracksToRemove) 2351{ 2352 2353 mixer_state mixerStatus = MIXER_IDLE; 2354 // find out which tracks need to be processed 2355 size_t count = mActiveTracks.size(); 2356 size_t mixedTracks = 0; 2357 size_t tracksWithEffect = 0; 2358 2359 float masterVolume = mMasterVolume; 2360 bool masterMute = mMasterMute; 2361 2362 if (masterMute) { 2363 masterVolume = 0; 2364 } 2365 // Delegate master volume control to effect in output mix effect chain if needed 2366 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2367 if (chain != 0) { 2368 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2369 chain->setVolume_l(&v, &v); 2370 masterVolume = (float)((v + (1 << 23)) >> 24); 2371 chain.clear(); 2372 } 2373 2374 for (size_t i=0 ; i<count ; i++) { 2375 sp<Track> t = mActiveTracks[i].promote(); 2376 if (t == 0) continue; 2377 2378 // this const just means the local variable doesn't change 2379 Track* const track = t.get(); 2380 audio_track_cblk_t* cblk = track->cblk(); 2381 2382 // The first time a track is added we wait 2383 // for all its buffers to be filled before processing it 2384 int name = track->name(); 2385 // make sure that we have enough frames to mix one full buffer. 2386 // enforce this condition only once to enable draining the buffer in case the client 2387 // app does not call stop() and relies on underrun to stop: 2388 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2389 // during last round 2390 uint32_t minFrames = 1; 2391 if (!track->isStopped() && !track->isPausing() && 2392 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2393 if (t->sampleRate() == (int)mSampleRate) { 2394 minFrames = mFrameCount; 2395 } else { 2396 // +1 for rounding and +1 for additional sample needed for interpolation 2397 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2398 // add frames already consumed but not yet released by the resampler 2399 // because cblk->framesReady() will include these frames 2400 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2401 // the minimum track buffer size is normally twice the number of frames necessary 2402 // to fill one buffer and the resampler should not leave more than one buffer worth 2403 // of unreleased frames after each pass, but just in case... 2404 ALOG_ASSERT(minFrames <= cblk->frameCount); 2405 } 2406 } 2407 if ((track->framesReady() >= minFrames) && track->isReady() && 2408 !track->isPaused() && !track->isTerminated()) 2409 { 2410 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2411 2412 mixedTracks++; 2413 2414 // track->mainBuffer() != mMixBuffer means there is an effect chain 2415 // connected to the track 2416 chain.clear(); 2417 if (track->mainBuffer() != mMixBuffer) { 2418 chain = getEffectChain_l(track->sessionId()); 2419 // Delegate volume control to effect in track effect chain if needed 2420 if (chain != 0) { 2421 tracksWithEffect++; 2422 } else { 2423 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2424 name, track->sessionId()); 2425 } 2426 } 2427 2428 2429 int param = AudioMixer::VOLUME; 2430 if (track->mFillingUpStatus == Track::FS_FILLED) { 2431 // no ramp for the first volume setting 2432 track->mFillingUpStatus = Track::FS_ACTIVE; 2433 if (track->mState == TrackBase::RESUMING) { 2434 track->mState = TrackBase::ACTIVE; 2435 param = AudioMixer::RAMP_VOLUME; 2436 } 2437 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2438 } else if (cblk->server != 0) { 2439 // If the track is stopped before the first frame was mixed, 2440 // do not apply ramp 2441 param = AudioMixer::RAMP_VOLUME; 2442 } 2443 2444 // compute volume for this track 2445 uint32_t vl, vr, va; 2446 if (track->isMuted() || track->isPausing() || 2447 mStreamTypes[track->streamType()].mute) { 2448 vl = vr = va = 0; 2449 if (track->isPausing()) { 2450 track->setPaused(); 2451 } 2452 } else { 2453 2454 // read original volumes with volume control 2455 float typeVolume = mStreamTypes[track->streamType()].volume; 2456 float v = masterVolume * typeVolume; 2457 uint32_t vlr = cblk->getVolumeLR(); 2458 vl = vlr & 0xFFFF; 2459 vr = vlr >> 16; 2460 // track volumes come from shared memory, so can't be trusted and must be clamped 2461 if (vl > MAX_GAIN_INT) { 2462 ALOGV("Track left volume out of range: %04X", vl); 2463 vl = MAX_GAIN_INT; 2464 } 2465 if (vr > MAX_GAIN_INT) { 2466 ALOGV("Track right volume out of range: %04X", vr); 2467 vr = MAX_GAIN_INT; 2468 } 2469 // now apply the master volume and stream type volume 2470 vl = (uint32_t)(v * vl) << 12; 2471 vr = (uint32_t)(v * vr) << 12; 2472 // assuming master volume and stream type volume each go up to 1.0, 2473 // vl and vr are now in 8.24 format 2474 2475 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2476 // send level comes from shared memory and so may be corrupt 2477 if (sendLevel > MAX_GAIN_INT) { 2478 ALOGV("Track send level out of range: %04X", sendLevel); 2479 sendLevel = MAX_GAIN_INT; 2480 } 2481 va = (uint32_t)(v * sendLevel); 2482 } 2483 // Delegate volume control to effect in track effect chain if needed 2484 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2485 // Do not ramp volume if volume is controlled by effect 2486 param = AudioMixer::VOLUME; 2487 track->mHasVolumeController = true; 2488 } else { 2489 // force no volume ramp when volume controller was just disabled or removed 2490 // from effect chain to avoid volume spike 2491 if (track->mHasVolumeController) { 2492 param = AudioMixer::VOLUME; 2493 } 2494 track->mHasVolumeController = false; 2495 } 2496 2497 // Convert volumes from 8.24 to 4.12 format 2498 // This additional clamping is needed in case chain->setVolume_l() overshot 2499 vl = (vl + (1 << 11)) >> 12; 2500 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2501 vr = (vr + (1 << 11)) >> 12; 2502 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2503 2504 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2505 2506 // XXX: these things DON'T need to be done each time 2507 mAudioMixer->setBufferProvider(name, track); 2508 mAudioMixer->enable(name); 2509 2510 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2511 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2512 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2513 mAudioMixer->setParameter( 2514 name, 2515 AudioMixer::TRACK, 2516 AudioMixer::FORMAT, (void *)track->format()); 2517 mAudioMixer->setParameter( 2518 name, 2519 AudioMixer::TRACK, 2520 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2521 mAudioMixer->setParameter( 2522 name, 2523 AudioMixer::RESAMPLE, 2524 AudioMixer::SAMPLE_RATE, 2525 (void *)(cblk->sampleRate)); 2526 mAudioMixer->setParameter( 2527 name, 2528 AudioMixer::TRACK, 2529 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2530 mAudioMixer->setParameter( 2531 name, 2532 AudioMixer::TRACK, 2533 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2534 2535 // reset retry count 2536 track->mRetryCount = kMaxTrackRetries; 2537 2538 // If one track is ready, set the mixer ready if: 2539 // - the mixer was not ready during previous round OR 2540 // - no other track is not ready 2541 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2542 mixerStatus != MIXER_TRACKS_ENABLED) { 2543 mixerStatus = MIXER_TRACKS_READY; 2544 } 2545 } else { 2546 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2547 if (track->isStopped()) { 2548 track->reset(); 2549 } 2550 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2551 // We have consumed all the buffers of this track. 2552 // Remove it from the list of active tracks. 2553 // TODO: use actual buffer filling status instead of latency when available from 2554 // audio HAL 2555 size_t audioHALFrames = 2556 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2557 size_t framesWritten = 2558 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2559 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2560 tracksToRemove->add(track); 2561 } 2562 } else { 2563 // No buffers for this track. Give it a few chances to 2564 // fill a buffer, then remove it from active list. 2565 if (--(track->mRetryCount) <= 0) { 2566 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2567 tracksToRemove->add(track); 2568 // indicate to client process that the track was disabled because of underrun 2569 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2570 // If one track is not ready, mark the mixer also not ready if: 2571 // - the mixer was ready during previous round OR 2572 // - no other track is ready 2573 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2574 mixerStatus != MIXER_TRACKS_READY) { 2575 mixerStatus = MIXER_TRACKS_ENABLED; 2576 } 2577 } 2578 mAudioMixer->disable(name); 2579 } 2580 } 2581 2582 // remove all the tracks that need to be... 2583 count = tracksToRemove->size(); 2584 if (CC_UNLIKELY(count)) { 2585 for (size_t i=0 ; i<count ; i++) { 2586 const sp<Track>& track = tracksToRemove->itemAt(i); 2587 mActiveTracks.remove(track); 2588 if (track->mainBuffer() != mMixBuffer) { 2589 chain = getEffectChain_l(track->sessionId()); 2590 if (chain != 0) { 2591 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2592 chain->decActiveTrackCnt(); 2593 } 2594 } 2595 if (track->isTerminated()) { 2596 removeTrack_l(track); 2597 } 2598 } 2599 } 2600 2601 // mix buffer must be cleared if all tracks are connected to an 2602 // effect chain as in this case the mixer will not write to 2603 // mix buffer and track effects will accumulate into it 2604 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2605 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2606 } 2607 2608 return mixerStatus; 2609} 2610 2611/* 2612The derived values that are cached: 2613 - mixBufferSize from frame count * frame size 2614 - activeSleepTime from activeSleepTimeUs() 2615 - idleSleepTime from idleSleepTimeUs() 2616 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2617 - maxPeriod from frame count and sample rate (MIXER only) 2618 2619The parameters that affect these derived values are: 2620 - frame count 2621 - frame size 2622 - sample rate 2623 - device type: A2DP or not 2624 - device latency 2625 - format: PCM or not 2626 - active sleep time 2627 - idle sleep time 2628*/ 2629 2630void AudioFlinger::PlaybackThread::cacheParameters_l() 2631{ 2632 mixBufferSize = mFrameCount * mFrameSize; 2633 activeSleepTime = activeSleepTimeUs(); 2634 idleSleepTime = idleSleepTimeUs(); 2635} 2636 2637void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2638{ 2639 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2640 this, streamType, mTracks.size()); 2641 Mutex::Autolock _l(mLock); 2642 2643 size_t size = mTracks.size(); 2644 for (size_t i = 0; i < size; i++) { 2645 sp<Track> t = mTracks[i]; 2646 if (t->streamType() == streamType) { 2647 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2648 t->mCblk->cv.signal(); 2649 } 2650 } 2651} 2652 2653// getTrackName_l() must be called with ThreadBase::mLock held 2654int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 2655{ 2656 int name = mAudioMixer->getTrackName(); 2657 if (name >= 0) { 2658 mAudioMixer->setParameter(name, 2659 AudioMixer::TRACK, 2660 AudioMixer::CHANNEL_MASK, (void *)channelMask); 2661 } 2662 return name; 2663} 2664 2665// deleteTrackName_l() must be called with ThreadBase::mLock held 2666void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2667{ 2668 ALOGV("remove track (%d) and delete from mixer", name); 2669 mAudioMixer->deleteTrackName(name); 2670} 2671 2672// checkForNewParameters_l() must be called with ThreadBase::mLock held 2673bool AudioFlinger::MixerThread::checkForNewParameters_l() 2674{ 2675 bool reconfig = false; 2676 2677 while (!mNewParameters.isEmpty()) { 2678 status_t status = NO_ERROR; 2679 String8 keyValuePair = mNewParameters[0]; 2680 AudioParameter param = AudioParameter(keyValuePair); 2681 int value; 2682 2683 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2684 reconfig = true; 2685 } 2686 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2687 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2688 status = BAD_VALUE; 2689 } else { 2690 reconfig = true; 2691 } 2692 } 2693 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2694 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2695 status = BAD_VALUE; 2696 } else { 2697 reconfig = true; 2698 } 2699 } 2700 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2701 // do not accept frame count changes if tracks are open as the track buffer 2702 // size depends on frame count and correct behavior would not be guaranteed 2703 // if frame count is changed after track creation 2704 if (!mTracks.isEmpty()) { 2705 status = INVALID_OPERATION; 2706 } else { 2707 reconfig = true; 2708 } 2709 } 2710 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2711#ifdef ADD_BATTERY_DATA 2712 // when changing the audio output device, call addBatteryData to notify 2713 // the change 2714 if ((int)mDevice != value) { 2715 uint32_t params = 0; 2716 // check whether speaker is on 2717 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2718 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2719 } 2720 2721 int deviceWithoutSpeaker 2722 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2723 // check if any other device (except speaker) is on 2724 if (value & deviceWithoutSpeaker ) { 2725 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2726 } 2727 2728 if (params != 0) { 2729 addBatteryData(params); 2730 } 2731 } 2732#endif 2733 2734 // forward device change to effects that have requested to be 2735 // aware of attached audio device. 2736 mDevice = (uint32_t)value; 2737 for (size_t i = 0; i < mEffectChains.size(); i++) { 2738 mEffectChains[i]->setDevice_l(mDevice); 2739 } 2740 } 2741 2742 if (status == NO_ERROR) { 2743 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2744 keyValuePair.string()); 2745 if (!mStandby && status == INVALID_OPERATION) { 2746 mOutput->stream->common.standby(&mOutput->stream->common); 2747 mStandby = true; 2748 mBytesWritten = 0; 2749 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2750 keyValuePair.string()); 2751 } 2752 if (status == NO_ERROR && reconfig) { 2753 delete mAudioMixer; 2754 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2755 mAudioMixer = NULL; 2756 readOutputParameters(); 2757 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2758 for (size_t i = 0; i < mTracks.size() ; i++) { 2759 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 2760 if (name < 0) break; 2761 mTracks[i]->mName = name; 2762 // limit track sample rate to 2 x new output sample rate 2763 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2764 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2765 } 2766 } 2767 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2768 } 2769 } 2770 2771 mNewParameters.removeAt(0); 2772 2773 mParamStatus = status; 2774 mParamCond.signal(); 2775 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2776 // already timed out waiting for the status and will never signal the condition. 2777 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2778 } 2779 return reconfig; 2780} 2781 2782status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2783{ 2784 const size_t SIZE = 256; 2785 char buffer[SIZE]; 2786 String8 result; 2787 2788 PlaybackThread::dumpInternals(fd, args); 2789 2790 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2791 result.append(buffer); 2792 write(fd, result.string(), result.size()); 2793 return NO_ERROR; 2794} 2795 2796uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 2797{ 2798 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2799} 2800 2801uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 2802{ 2803 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2804} 2805 2806void AudioFlinger::MixerThread::cacheParameters_l() 2807{ 2808 PlaybackThread::cacheParameters_l(); 2809 2810 // FIXME: Relaxed timing because of a certain device that can't meet latency 2811 // Should be reduced to 2x after the vendor fixes the driver issue 2812 // increase threshold again due to low power audio mode. The way this warning 2813 // threshold is calculated and its usefulness should be reconsidered anyway. 2814 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2815} 2816 2817// ---------------------------------------------------------------------------- 2818AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2819 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2820 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2821 // mLeftVolFloat, mRightVolFloat 2822 // mLeftVolShort, mRightVolShort 2823{ 2824} 2825 2826AudioFlinger::DirectOutputThread::~DirectOutputThread() 2827{ 2828} 2829 2830AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2831 Vector< sp<Track> > *tracksToRemove 2832) 2833{ 2834 sp<Track> trackToRemove; 2835 2836 mixer_state mixerStatus = MIXER_IDLE; 2837 2838 // find out which tracks need to be processed 2839 if (mActiveTracks.size() != 0) { 2840 sp<Track> t = mActiveTracks[0].promote(); 2841 // The track died recently 2842 if (t == 0) return MIXER_IDLE; 2843 2844 Track* const track = t.get(); 2845 audio_track_cblk_t* cblk = track->cblk(); 2846 2847 // The first time a track is added we wait 2848 // for all its buffers to be filled before processing it 2849 if (cblk->framesReady() && track->isReady() && 2850 !track->isPaused() && !track->isTerminated()) 2851 { 2852 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2853 2854 if (track->mFillingUpStatus == Track::FS_FILLED) { 2855 track->mFillingUpStatus = Track::FS_ACTIVE; 2856 mLeftVolFloat = mRightVolFloat = 0; 2857 mLeftVolShort = mRightVolShort = 0; 2858 if (track->mState == TrackBase::RESUMING) { 2859 track->mState = TrackBase::ACTIVE; 2860 rampVolume = true; 2861 } 2862 } else if (cblk->server != 0) { 2863 // If the track is stopped before the first frame was mixed, 2864 // do not apply ramp 2865 rampVolume = true; 2866 } 2867 // compute volume for this track 2868 float left, right; 2869 if (track->isMuted() || mMasterMute || track->isPausing() || 2870 mStreamTypes[track->streamType()].mute) { 2871 left = right = 0; 2872 if (track->isPausing()) { 2873 track->setPaused(); 2874 } 2875 } else { 2876 float typeVolume = mStreamTypes[track->streamType()].volume; 2877 float v = mMasterVolume * typeVolume; 2878 uint32_t vlr = cblk->getVolumeLR(); 2879 float v_clamped = v * (vlr & 0xFFFF); 2880 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2881 left = v_clamped/MAX_GAIN; 2882 v_clamped = v * (vlr >> 16); 2883 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2884 right = v_clamped/MAX_GAIN; 2885 } 2886 2887 if (left != mLeftVolFloat || right != mRightVolFloat) { 2888 mLeftVolFloat = left; 2889 mRightVolFloat = right; 2890 2891 // If audio HAL implements volume control, 2892 // force software volume to nominal value 2893 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2894 left = 1.0f; 2895 right = 1.0f; 2896 } 2897 2898 // Convert volumes from float to 8.24 2899 uint32_t vl = (uint32_t)(left * (1 << 24)); 2900 uint32_t vr = (uint32_t)(right * (1 << 24)); 2901 2902 // Delegate volume control to effect in track effect chain if needed 2903 // only one effect chain can be present on DirectOutputThread, so if 2904 // there is one, the track is connected to it 2905 if (!mEffectChains.isEmpty()) { 2906 // Do not ramp volume if volume is controlled by effect 2907 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2908 rampVolume = false; 2909 } 2910 } 2911 2912 // Convert volumes from 8.24 to 4.12 format 2913 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2914 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2915 leftVol = (uint16_t)v_clamped; 2916 v_clamped = (vr + (1 << 11)) >> 12; 2917 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2918 rightVol = (uint16_t)v_clamped; 2919 } else { 2920 leftVol = mLeftVolShort; 2921 rightVol = mRightVolShort; 2922 rampVolume = false; 2923 } 2924 2925 // reset retry count 2926 track->mRetryCount = kMaxTrackRetriesDirect; 2927 mActiveTrack = t; 2928 mixerStatus = MIXER_TRACKS_READY; 2929 } else { 2930 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2931 if (track->isStopped()) { 2932 track->reset(); 2933 } 2934 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2935 // We have consumed all the buffers of this track. 2936 // Remove it from the list of active tracks. 2937 // TODO: implement behavior for compressed audio 2938 size_t audioHALFrames = 2939 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2940 size_t framesWritten = 2941 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2942 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2943 trackToRemove = track; 2944 } 2945 } else { 2946 // No buffers for this track. Give it a few chances to 2947 // fill a buffer, then remove it from active list. 2948 if (--(track->mRetryCount) <= 0) { 2949 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2950 trackToRemove = track; 2951 } else { 2952 mixerStatus = MIXER_TRACKS_ENABLED; 2953 } 2954 } 2955 } 2956 } 2957 2958 // FIXME merge this with similar code for removing multiple tracks 2959 // remove all the tracks that need to be... 2960 if (CC_UNLIKELY(trackToRemove != 0)) { 2961 tracksToRemove->add(trackToRemove); 2962 mActiveTracks.remove(trackToRemove); 2963 if (!mEffectChains.isEmpty()) { 2964 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2965 trackToRemove->sessionId()); 2966 mEffectChains[0]->decActiveTrackCnt(); 2967 } 2968 if (trackToRemove->isTerminated()) { 2969 removeTrack_l(trackToRemove); 2970 } 2971 } 2972 2973 return mixerStatus; 2974} 2975 2976void AudioFlinger::DirectOutputThread::threadLoop_mix() 2977{ 2978 AudioBufferProvider::Buffer buffer; 2979 size_t frameCount = mFrameCount; 2980 int8_t *curBuf = (int8_t *)mMixBuffer; 2981 // output audio to hardware 2982 while (frameCount) { 2983 buffer.frameCount = frameCount; 2984 mActiveTrack->getNextBuffer(&buffer); 2985 if (CC_UNLIKELY(buffer.raw == NULL)) { 2986 memset(curBuf, 0, frameCount * mFrameSize); 2987 break; 2988 } 2989 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2990 frameCount -= buffer.frameCount; 2991 curBuf += buffer.frameCount * mFrameSize; 2992 mActiveTrack->releaseBuffer(&buffer); 2993 } 2994 sleepTime = 0; 2995 standbyTime = systemTime() + standbyDelay; 2996 mActiveTrack.clear(); 2997 2998 // apply volume 2999 3000 // Do not apply volume on compressed audio 3001 if (!audio_is_linear_pcm(mFormat)) { 3002 return; 3003 } 3004 3005 // convert to signed 16 bit before volume calculation 3006 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3007 size_t count = mFrameCount * mChannelCount; 3008 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3009 int16_t *dst = mMixBuffer + count-1; 3010 while (count--) { 3011 *dst-- = (int16_t)(*src--^0x80) << 8; 3012 } 3013 } 3014 3015 frameCount = mFrameCount; 3016 int16_t *out = mMixBuffer; 3017 if (rampVolume) { 3018 if (mChannelCount == 1) { 3019 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3020 int32_t vlInc = d / (int32_t)frameCount; 3021 int32_t vl = ((int32_t)mLeftVolShort << 16); 3022 do { 3023 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3024 out++; 3025 vl += vlInc; 3026 } while (--frameCount); 3027 3028 } else { 3029 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3030 int32_t vlInc = d / (int32_t)frameCount; 3031 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3032 int32_t vrInc = d / (int32_t)frameCount; 3033 int32_t vl = ((int32_t)mLeftVolShort << 16); 3034 int32_t vr = ((int32_t)mRightVolShort << 16); 3035 do { 3036 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3037 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3038 out += 2; 3039 vl += vlInc; 3040 vr += vrInc; 3041 } while (--frameCount); 3042 } 3043 } else { 3044 if (mChannelCount == 1) { 3045 do { 3046 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3047 out++; 3048 } while (--frameCount); 3049 } else { 3050 do { 3051 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3052 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3053 out += 2; 3054 } while (--frameCount); 3055 } 3056 } 3057 3058 // convert back to unsigned 8 bit after volume calculation 3059 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3060 size_t count = mFrameCount * mChannelCount; 3061 int16_t *src = mMixBuffer; 3062 uint8_t *dst = (uint8_t *)mMixBuffer; 3063 while (count--) { 3064 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3065 } 3066 } 3067 3068 mLeftVolShort = leftVol; 3069 mRightVolShort = rightVol; 3070} 3071 3072void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3073{ 3074 if (sleepTime == 0) { 3075 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3076 sleepTime = activeSleepTime; 3077 } else { 3078 sleepTime = idleSleepTime; 3079 } 3080 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3081 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3082 sleepTime = 0; 3083 } 3084} 3085 3086// getTrackName_l() must be called with ThreadBase::mLock held 3087int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3088{ 3089 return 0; 3090} 3091 3092// deleteTrackName_l() must be called with ThreadBase::mLock held 3093void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3094{ 3095} 3096 3097// checkForNewParameters_l() must be called with ThreadBase::mLock held 3098bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3099{ 3100 bool reconfig = false; 3101 3102 while (!mNewParameters.isEmpty()) { 3103 status_t status = NO_ERROR; 3104 String8 keyValuePair = mNewParameters[0]; 3105 AudioParameter param = AudioParameter(keyValuePair); 3106 int value; 3107 3108 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3109 // do not accept frame count changes if tracks are open as the track buffer 3110 // size depends on frame count and correct behavior would not be garantied 3111 // if frame count is changed after track creation 3112 if (!mTracks.isEmpty()) { 3113 status = INVALID_OPERATION; 3114 } else { 3115 reconfig = true; 3116 } 3117 } 3118 if (status == NO_ERROR) { 3119 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3120 keyValuePair.string()); 3121 if (!mStandby && status == INVALID_OPERATION) { 3122 mOutput->stream->common.standby(&mOutput->stream->common); 3123 mStandby = true; 3124 mBytesWritten = 0; 3125 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3126 keyValuePair.string()); 3127 } 3128 if (status == NO_ERROR && reconfig) { 3129 readOutputParameters(); 3130 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3131 } 3132 } 3133 3134 mNewParameters.removeAt(0); 3135 3136 mParamStatus = status; 3137 mParamCond.signal(); 3138 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3139 // already timed out waiting for the status and will never signal the condition. 3140 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3141 } 3142 return reconfig; 3143} 3144 3145uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3146{ 3147 uint32_t time; 3148 if (audio_is_linear_pcm(mFormat)) { 3149 time = PlaybackThread::activeSleepTimeUs(); 3150 } else { 3151 time = 10000; 3152 } 3153 return time; 3154} 3155 3156uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3157{ 3158 uint32_t time; 3159 if (audio_is_linear_pcm(mFormat)) { 3160 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3161 } else { 3162 time = 10000; 3163 } 3164 return time; 3165} 3166 3167uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3168{ 3169 uint32_t time; 3170 if (audio_is_linear_pcm(mFormat)) { 3171 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3172 } else { 3173 time = 10000; 3174 } 3175 return time; 3176} 3177 3178void AudioFlinger::DirectOutputThread::cacheParameters_l() 3179{ 3180 PlaybackThread::cacheParameters_l(); 3181 3182 // use shorter standby delay as on normal output to release 3183 // hardware resources as soon as possible 3184 standbyDelay = microseconds(activeSleepTime*2); 3185} 3186 3187// ---------------------------------------------------------------------------- 3188 3189AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3190 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3191 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3192 mWaitTimeMs(UINT_MAX) 3193{ 3194 addOutputTrack(mainThread); 3195} 3196 3197AudioFlinger::DuplicatingThread::~DuplicatingThread() 3198{ 3199 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3200 mOutputTracks[i]->destroy(); 3201 } 3202} 3203 3204void AudioFlinger::DuplicatingThread::threadLoop_mix() 3205{ 3206 // mix buffers... 3207 if (outputsReady(outputTracks)) { 3208 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3209 } else { 3210 memset(mMixBuffer, 0, mixBufferSize); 3211 } 3212 sleepTime = 0; 3213 writeFrames = mFrameCount; 3214} 3215 3216void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3217{ 3218 if (sleepTime == 0) { 3219 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3220 sleepTime = activeSleepTime; 3221 } else { 3222 sleepTime = idleSleepTime; 3223 } 3224 } else if (mBytesWritten != 0) { 3225 // flush remaining overflow buffers in output tracks 3226 for (size_t i = 0; i < outputTracks.size(); i++) { 3227 if (outputTracks[i]->isActive()) { 3228 sleepTime = 0; 3229 writeFrames = 0; 3230 memset(mMixBuffer, 0, mixBufferSize); 3231 break; 3232 } 3233 } 3234 } 3235} 3236 3237void AudioFlinger::DuplicatingThread::threadLoop_write() 3238{ 3239 standbyTime = systemTime() + standbyDelay; 3240 for (size_t i = 0; i < outputTracks.size(); i++) { 3241 outputTracks[i]->write(mMixBuffer, writeFrames); 3242 } 3243 mBytesWritten += mixBufferSize; 3244} 3245 3246void AudioFlinger::DuplicatingThread::threadLoop_standby() 3247{ 3248 // DuplicatingThread implements standby by stopping all tracks 3249 for (size_t i = 0; i < outputTracks.size(); i++) { 3250 outputTracks[i]->stop(); 3251 } 3252} 3253 3254void AudioFlinger::DuplicatingThread::saveOutputTracks() 3255{ 3256 outputTracks = mOutputTracks; 3257} 3258 3259void AudioFlinger::DuplicatingThread::clearOutputTracks() 3260{ 3261 outputTracks.clear(); 3262} 3263 3264void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3265{ 3266 Mutex::Autolock _l(mLock); 3267 // FIXME explain this formula 3268 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3269 OutputTrack *outputTrack = new OutputTrack(thread, 3270 this, 3271 mSampleRate, 3272 mFormat, 3273 mChannelMask, 3274 frameCount); 3275 if (outputTrack->cblk() != NULL) { 3276 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3277 mOutputTracks.add(outputTrack); 3278 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3279 updateWaitTime_l(); 3280 } 3281} 3282 3283void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3284{ 3285 Mutex::Autolock _l(mLock); 3286 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3287 if (mOutputTracks[i]->thread() == thread) { 3288 mOutputTracks[i]->destroy(); 3289 mOutputTracks.removeAt(i); 3290 updateWaitTime_l(); 3291 return; 3292 } 3293 } 3294 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3295} 3296 3297// caller must hold mLock 3298void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3299{ 3300 mWaitTimeMs = UINT_MAX; 3301 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3302 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3303 if (strong != 0) { 3304 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3305 if (waitTimeMs < mWaitTimeMs) { 3306 mWaitTimeMs = waitTimeMs; 3307 } 3308 } 3309 } 3310} 3311 3312 3313bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3314{ 3315 for (size_t i = 0; i < outputTracks.size(); i++) { 3316 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3317 if (thread == 0) { 3318 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3319 return false; 3320 } 3321 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3322 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3323 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3324 return false; 3325 } 3326 } 3327 return true; 3328} 3329 3330uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3331{ 3332 return (mWaitTimeMs * 1000) / 2; 3333} 3334 3335void AudioFlinger::DuplicatingThread::cacheParameters_l() 3336{ 3337 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3338 updateWaitTime_l(); 3339 3340 MixerThread::cacheParameters_l(); 3341} 3342 3343// ---------------------------------------------------------------------------- 3344 3345// TrackBase constructor must be called with AudioFlinger::mLock held 3346AudioFlinger::ThreadBase::TrackBase::TrackBase( 3347 ThreadBase *thread, 3348 const sp<Client>& client, 3349 uint32_t sampleRate, 3350 audio_format_t format, 3351 uint32_t channelMask, 3352 int frameCount, 3353 const sp<IMemory>& sharedBuffer, 3354 int sessionId) 3355 : RefBase(), 3356 mThread(thread), 3357 mClient(client), 3358 mCblk(NULL), 3359 // mBuffer 3360 // mBufferEnd 3361 mFrameCount(0), 3362 mState(IDLE), 3363 mFormat(format), 3364 mStepServerFailed(false), 3365 mSessionId(sessionId) 3366 // mChannelCount 3367 // mChannelMask 3368{ 3369 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3370 3371 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3372 size_t size = sizeof(audio_track_cblk_t); 3373 uint8_t channelCount = popcount(channelMask); 3374 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3375 if (sharedBuffer == 0) { 3376 size += bufferSize; 3377 } 3378 3379 if (client != NULL) { 3380 mCblkMemory = client->heap()->allocate(size); 3381 if (mCblkMemory != 0) { 3382 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3383 if (mCblk != NULL) { // construct the shared structure in-place. 3384 new(mCblk) audio_track_cblk_t(); 3385 // clear all buffers 3386 mCblk->frameCount = frameCount; 3387 mCblk->sampleRate = sampleRate; 3388// uncomment the following lines to quickly test 32-bit wraparound 3389// mCblk->user = 0xffff0000; 3390// mCblk->server = 0xffff0000; 3391// mCblk->userBase = 0xffff0000; 3392// mCblk->serverBase = 0xffff0000; 3393 mChannelCount = channelCount; 3394 mChannelMask = channelMask; 3395 if (sharedBuffer == 0) { 3396 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3397 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3398 // Force underrun condition to avoid false underrun callback until first data is 3399 // written to buffer (other flags are cleared) 3400 mCblk->flags = CBLK_UNDERRUN_ON; 3401 } else { 3402 mBuffer = sharedBuffer->pointer(); 3403 } 3404 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3405 } 3406 } else { 3407 ALOGE("not enough memory for AudioTrack size=%u", size); 3408 client->heap()->dump("AudioTrack"); 3409 return; 3410 } 3411 } else { 3412 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3413 // construct the shared structure in-place. 3414 new(mCblk) audio_track_cblk_t(); 3415 // clear all buffers 3416 mCblk->frameCount = frameCount; 3417 mCblk->sampleRate = sampleRate; 3418// uncomment the following lines to quickly test 32-bit wraparound 3419// mCblk->user = 0xffff0000; 3420// mCblk->server = 0xffff0000; 3421// mCblk->userBase = 0xffff0000; 3422// mCblk->serverBase = 0xffff0000; 3423 mChannelCount = channelCount; 3424 mChannelMask = channelMask; 3425 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3426 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3427 // Force underrun condition to avoid false underrun callback until first data is 3428 // written to buffer (other flags are cleared) 3429 mCblk->flags = CBLK_UNDERRUN_ON; 3430 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3431 } 3432} 3433 3434AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3435{ 3436 if (mCblk != NULL) { 3437 if (mClient == 0) { 3438 delete mCblk; 3439 } else { 3440 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3441 } 3442 } 3443 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3444 if (mClient != 0) { 3445 // Client destructor must run with AudioFlinger mutex locked 3446 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3447 // If the client's reference count drops to zero, the associated destructor 3448 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3449 // relying on the automatic clear() at end of scope. 3450 mClient.clear(); 3451 } 3452} 3453 3454// AudioBufferProvider interface 3455// getNextBuffer() = 0; 3456// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3457void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3458{ 3459 buffer->raw = NULL; 3460 mFrameCount = buffer->frameCount; 3461 (void) step(); // ignore return value of step() 3462 buffer->frameCount = 0; 3463} 3464 3465bool AudioFlinger::ThreadBase::TrackBase::step() { 3466 bool result; 3467 audio_track_cblk_t* cblk = this->cblk(); 3468 3469 result = cblk->stepServer(mFrameCount); 3470 if (!result) { 3471 ALOGV("stepServer failed acquiring cblk mutex"); 3472 mStepServerFailed = true; 3473 } 3474 return result; 3475} 3476 3477void AudioFlinger::ThreadBase::TrackBase::reset() { 3478 audio_track_cblk_t* cblk = this->cblk(); 3479 3480 cblk->user = 0; 3481 cblk->server = 0; 3482 cblk->userBase = 0; 3483 cblk->serverBase = 0; 3484 mStepServerFailed = false; 3485 ALOGV("TrackBase::reset"); 3486} 3487 3488int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3489 return (int)mCblk->sampleRate; 3490} 3491 3492void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3493 audio_track_cblk_t* cblk = this->cblk(); 3494 size_t frameSize = cblk->frameSize; 3495 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3496 int8_t *bufferEnd = bufferStart + frames * frameSize; 3497 3498 // Check validity of returned pointer in case the track control block would have been corrupted. 3499 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3500 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3501 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3502 server %u, serverBase %u, user %u, userBase %u", 3503 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3504 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3505 return NULL; 3506 } 3507 3508 return bufferStart; 3509} 3510 3511status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3512{ 3513 mSyncEvents.add(event); 3514 return NO_ERROR; 3515} 3516 3517// ---------------------------------------------------------------------------- 3518 3519// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3520AudioFlinger::PlaybackThread::Track::Track( 3521 PlaybackThread *thread, 3522 const sp<Client>& client, 3523 audio_stream_type_t streamType, 3524 uint32_t sampleRate, 3525 audio_format_t format, 3526 uint32_t channelMask, 3527 int frameCount, 3528 const sp<IMemory>& sharedBuffer, 3529 int sessionId, 3530 IAudioFlinger::track_flags_t flags) 3531 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3532 mMute(false), 3533 // mFillingUpStatus ? 3534 // mRetryCount initialized later when needed 3535 mSharedBuffer(sharedBuffer), 3536 mStreamType(streamType), 3537 mName(-1), // see note below 3538 mMainBuffer(thread->mixBuffer()), 3539 mAuxBuffer(NULL), 3540 mAuxEffectId(0), mHasVolumeController(false), 3541 mPresentationCompleteFrames(0), 3542 mFlags(flags) 3543{ 3544 if (mCblk != NULL) { 3545 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3546 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3547 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3548 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3549 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 3550 if (mName < 0) { 3551 ALOGE("no more track names available"); 3552 } 3553 } 3554 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3555} 3556 3557AudioFlinger::PlaybackThread::Track::~Track() 3558{ 3559 ALOGV("PlaybackThread::Track destructor"); 3560 sp<ThreadBase> thread = mThread.promote(); 3561 if (thread != 0) { 3562 Mutex::Autolock _l(thread->mLock); 3563 mState = TERMINATED; 3564 } 3565} 3566 3567void AudioFlinger::PlaybackThread::Track::destroy() 3568{ 3569 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3570 // by removing it from mTracks vector, so there is a risk that this Tracks's 3571 // destructor is called. As the destructor needs to lock mLock, 3572 // we must acquire a strong reference on this Track before locking mLock 3573 // here so that the destructor is called only when exiting this function. 3574 // On the other hand, as long as Track::destroy() is only called by 3575 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3576 // this Track with its member mTrack. 3577 sp<Track> keep(this); 3578 { // scope for mLock 3579 sp<ThreadBase> thread = mThread.promote(); 3580 if (thread != 0) { 3581 if (!isOutputTrack()) { 3582 if (mState == ACTIVE || mState == RESUMING) { 3583 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3584 3585#ifdef ADD_BATTERY_DATA 3586 // to track the speaker usage 3587 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3588#endif 3589 } 3590 AudioSystem::releaseOutput(thread->id()); 3591 } 3592 Mutex::Autolock _l(thread->mLock); 3593 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3594 playbackThread->destroyTrack_l(this); 3595 } 3596 } 3597} 3598 3599void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3600{ 3601 uint32_t vlr = mCblk->getVolumeLR(); 3602 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", 3603 mName - AudioMixer::TRACK0, 3604 (mClient == 0) ? getpid_cached : mClient->pid(), 3605 mStreamType, 3606 mFormat, 3607 mChannelMask, 3608 mSessionId, 3609 mFrameCount, 3610 mState, 3611 mMute, 3612 mFillingUpStatus, 3613 mCblk->sampleRate, 3614 vlr & 0xFFFF, 3615 vlr >> 16, 3616 mCblk->server, 3617 mCblk->user, 3618 (int)mMainBuffer, 3619 (int)mAuxBuffer); 3620} 3621 3622// AudioBufferProvider interface 3623status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3624 AudioBufferProvider::Buffer* buffer, int64_t pts) 3625{ 3626 audio_track_cblk_t* cblk = this->cblk(); 3627 uint32_t framesReady; 3628 uint32_t framesReq = buffer->frameCount; 3629 3630 // Check if last stepServer failed, try to step now 3631 if (mStepServerFailed) { 3632 if (!step()) goto getNextBuffer_exit; 3633 ALOGV("stepServer recovered"); 3634 mStepServerFailed = false; 3635 } 3636 3637 framesReady = cblk->framesReady(); 3638 3639 if (CC_LIKELY(framesReady)) { 3640 uint32_t s = cblk->server; 3641 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3642 3643 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3644 if (framesReq > framesReady) { 3645 framesReq = framesReady; 3646 } 3647 if (framesReq > bufferEnd - s) { 3648 framesReq = bufferEnd - s; 3649 } 3650 3651 buffer->raw = getBuffer(s, framesReq); 3652 if (buffer->raw == NULL) goto getNextBuffer_exit; 3653 3654 buffer->frameCount = framesReq; 3655 return NO_ERROR; 3656 } 3657 3658getNextBuffer_exit: 3659 buffer->raw = NULL; 3660 buffer->frameCount = 0; 3661 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3662 return NOT_ENOUGH_DATA; 3663} 3664 3665uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3666 return mCblk->framesReady(); 3667} 3668 3669bool AudioFlinger::PlaybackThread::Track::isReady() const { 3670 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3671 3672 if (framesReady() >= mCblk->frameCount || 3673 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3674 mFillingUpStatus = FS_FILLED; 3675 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3676 return true; 3677 } 3678 return false; 3679} 3680 3681status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid, 3682 AudioSystem::sync_event_t event, 3683 int triggerSession) 3684{ 3685 status_t status = NO_ERROR; 3686 ALOGV("start(%d), calling pid %d session %d tid %d", 3687 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3688 // check for use case 2 with missing callback 3689 if (isFastTrack() && (mSharedBuffer == 0) && (tid == 0)) { 3690 ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied"); 3691 mFlags &= ~IAudioFlinger::TRACK_FAST; 3692 // FIXME the track must be invalidated and moved to another thread or 3693 // attached directly to the normal mixer now 3694 } 3695 sp<ThreadBase> thread = mThread.promote(); 3696 if (thread != 0) { 3697 Mutex::Autolock _l(thread->mLock); 3698 track_state state = mState; 3699 // here the track could be either new, or restarted 3700 // in both cases "unstop" the track 3701 if (mState == PAUSED) { 3702 mState = TrackBase::RESUMING; 3703 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3704 } else { 3705 mState = TrackBase::ACTIVE; 3706 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3707 } 3708 3709 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3710 thread->mLock.unlock(); 3711 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3712 thread->mLock.lock(); 3713 3714#ifdef ADD_BATTERY_DATA 3715 // to track the speaker usage 3716 if (status == NO_ERROR) { 3717 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3718 } 3719#endif 3720 } 3721 if (status == NO_ERROR) { 3722 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3723 playbackThread->addTrack_l(this); 3724 } else { 3725 mState = state; 3726 } 3727 } else { 3728 status = BAD_VALUE; 3729 } 3730 return status; 3731} 3732 3733void AudioFlinger::PlaybackThread::Track::stop() 3734{ 3735 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3736 sp<ThreadBase> thread = mThread.promote(); 3737 if (thread != 0) { 3738 Mutex::Autolock _l(thread->mLock); 3739 track_state state = mState; 3740 if (mState > STOPPED) { 3741 mState = STOPPED; 3742 // If the track is not active (PAUSED and buffers full), flush buffers 3743 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3744 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3745 reset(); 3746 } 3747 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3748 } 3749 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3750 thread->mLock.unlock(); 3751 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3752 thread->mLock.lock(); 3753 3754#ifdef ADD_BATTERY_DATA 3755 // to track the speaker usage 3756 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3757#endif 3758 } 3759 } 3760} 3761 3762void AudioFlinger::PlaybackThread::Track::pause() 3763{ 3764 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3765 sp<ThreadBase> thread = mThread.promote(); 3766 if (thread != 0) { 3767 Mutex::Autolock _l(thread->mLock); 3768 if (mState == ACTIVE || mState == RESUMING) { 3769 mState = PAUSING; 3770 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3771 if (!isOutputTrack()) { 3772 thread->mLock.unlock(); 3773 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3774 thread->mLock.lock(); 3775 3776#ifdef ADD_BATTERY_DATA 3777 // to track the speaker usage 3778 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3779#endif 3780 } 3781 } 3782 } 3783} 3784 3785void AudioFlinger::PlaybackThread::Track::flush() 3786{ 3787 ALOGV("flush(%d)", mName); 3788 sp<ThreadBase> thread = mThread.promote(); 3789 if (thread != 0) { 3790 Mutex::Autolock _l(thread->mLock); 3791 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3792 return; 3793 } 3794 // No point remaining in PAUSED state after a flush => go to 3795 // STOPPED state 3796 mState = STOPPED; 3797 3798 // do not reset the track if it is still in the process of being stopped or paused. 3799 // this will be done by prepareTracks_l() when the track is stopped. 3800 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3801 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3802 reset(); 3803 } 3804 } 3805} 3806 3807void AudioFlinger::PlaybackThread::Track::reset() 3808{ 3809 // Do not reset twice to avoid discarding data written just after a flush and before 3810 // the audioflinger thread detects the track is stopped. 3811 if (!mResetDone) { 3812 TrackBase::reset(); 3813 // Force underrun condition to avoid false underrun callback until first data is 3814 // written to buffer 3815 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3816 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3817 mFillingUpStatus = FS_FILLING; 3818 mResetDone = true; 3819 mPresentationCompleteFrames = 0; 3820 } 3821} 3822 3823void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3824{ 3825 mMute = muted; 3826} 3827 3828status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3829{ 3830 status_t status = DEAD_OBJECT; 3831 sp<ThreadBase> thread = mThread.promote(); 3832 if (thread != 0) { 3833 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3834 status = playbackThread->attachAuxEffect(this, EffectId); 3835 } 3836 return status; 3837} 3838 3839void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3840{ 3841 mAuxEffectId = EffectId; 3842 mAuxBuffer = buffer; 3843} 3844 3845bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 3846 size_t audioHalFrames) 3847{ 3848 // a track is considered presented when the total number of frames written to audio HAL 3849 // corresponds to the number of frames written when presentationComplete() is called for the 3850 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 3851 if (mPresentationCompleteFrames == 0) { 3852 mPresentationCompleteFrames = framesWritten + audioHalFrames; 3853 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 3854 mPresentationCompleteFrames, audioHalFrames); 3855 } 3856 if (framesWritten >= mPresentationCompleteFrames) { 3857 ALOGV("presentationComplete() session %d complete: framesWritten %d", 3858 mSessionId, framesWritten); 3859 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 3860 mPresentationCompleteFrames = 0; 3861 return true; 3862 } 3863 return false; 3864} 3865 3866void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 3867{ 3868 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 3869 if (mSyncEvents[i]->type() == type) { 3870 mSyncEvents[i]->trigger(); 3871 mSyncEvents.removeAt(i); 3872 i--; 3873 } 3874 } 3875} 3876 3877 3878// timed audio tracks 3879 3880sp<AudioFlinger::PlaybackThread::TimedTrack> 3881AudioFlinger::PlaybackThread::TimedTrack::create( 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 if (!client->reserveTimedTrack()) 3892 return NULL; 3893 3894 return new TimedTrack( 3895 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3896 sharedBuffer, sessionId); 3897} 3898 3899AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3900 PlaybackThread *thread, 3901 const sp<Client>& client, 3902 audio_stream_type_t streamType, 3903 uint32_t sampleRate, 3904 audio_format_t format, 3905 uint32_t channelMask, 3906 int frameCount, 3907 const sp<IMemory>& sharedBuffer, 3908 int sessionId) 3909 : Track(thread, client, streamType, sampleRate, format, channelMask, 3910 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 3911 mQueueHeadInFlight(false), 3912 mTrimQueueHeadOnRelease(false), 3913 mFramesPendingInQueue(0), 3914 mTimedSilenceBuffer(NULL), 3915 mTimedSilenceBufferSize(0), 3916 mTimedAudioOutputOnTime(false), 3917 mMediaTimeTransformValid(false) 3918{ 3919 LocalClock lc; 3920 mLocalTimeFreq = lc.getLocalFreq(); 3921 3922 mLocalTimeToSampleTransform.a_zero = 0; 3923 mLocalTimeToSampleTransform.b_zero = 0; 3924 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3925 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3926 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3927 &mLocalTimeToSampleTransform.a_to_b_denom); 3928 3929 mMediaTimeToSampleTransform.a_zero = 0; 3930 mMediaTimeToSampleTransform.b_zero = 0; 3931 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 3932 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 3933 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 3934 &mMediaTimeToSampleTransform.a_to_b_denom); 3935} 3936 3937AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3938 mClient->releaseTimedTrack(); 3939 delete [] mTimedSilenceBuffer; 3940} 3941 3942status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3943 size_t size, sp<IMemory>* buffer) { 3944 3945 Mutex::Autolock _l(mTimedBufferQueueLock); 3946 3947 trimTimedBufferQueue_l(); 3948 3949 // lazily initialize the shared memory heap for timed buffers 3950 if (mTimedMemoryDealer == NULL) { 3951 const int kTimedBufferHeapSize = 512 << 10; 3952 3953 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3954 "AudioFlingerTimed"); 3955 if (mTimedMemoryDealer == NULL) 3956 return NO_MEMORY; 3957 } 3958 3959 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3960 if (newBuffer == NULL) { 3961 newBuffer = mTimedMemoryDealer->allocate(size); 3962 if (newBuffer == NULL) 3963 return NO_MEMORY; 3964 } 3965 3966 *buffer = newBuffer; 3967 return NO_ERROR; 3968} 3969 3970// caller must hold mTimedBufferQueueLock 3971void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3972 int64_t mediaTimeNow; 3973 { 3974 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3975 if (!mMediaTimeTransformValid) 3976 return; 3977 3978 int64_t targetTimeNow; 3979 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3980 ? mCCHelper.getCommonTime(&targetTimeNow) 3981 : mCCHelper.getLocalTime(&targetTimeNow); 3982 3983 if (OK != res) 3984 return; 3985 3986 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3987 &mediaTimeNow)) { 3988 return; 3989 } 3990 } 3991 3992 size_t trimEnd; 3993 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 3994 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 3995 / mCblk->frameSize; 3996 int64_t bufEnd; 3997 3998 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 3999 &bufEnd)) { 4000 ALOGE("Failed to convert frame count of %lld to media time duration" 4001 " (scale factor %d/%u) in %s", frameCount, 4002 mMediaTimeToSampleTransform.a_to_b_numer, 4003 mMediaTimeToSampleTransform.a_to_b_denom, 4004 __PRETTY_FUNCTION__); 4005 break; 4006 } 4007 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4008 4009 if (bufEnd > mediaTimeNow) 4010 break; 4011 4012 // Is the buffer we want to use in the middle of a mix operation right 4013 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4014 // from the mixer which should be coming back shortly. 4015 if (!trimEnd && mQueueHeadInFlight) { 4016 mTrimQueueHeadOnRelease = true; 4017 } 4018 } 4019 4020 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4021 if (trimStart < trimEnd) { 4022 // Update the bookkeeping for framesReady() 4023 for (size_t i = trimStart; i < trimEnd; ++i) { 4024 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4025 } 4026 4027 // Now actually remove the buffers from the queue. 4028 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4029 } 4030} 4031 4032void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4033 const char* logTag) { 4034 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4035 "%s called (reason \"%s\"), but timed buffer queue has no" 4036 " elements to trim.", __FUNCTION__, logTag); 4037 4038 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4039 mTimedBufferQueue.removeAt(0); 4040} 4041 4042void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4043 const TimedBuffer& buf, 4044 const char* logTag) { 4045 uint32_t bufBytes = buf.buffer()->size(); 4046 uint32_t consumedAlready = buf.position(); 4047 4048 ALOG_ASSERT(consumedAlready <= bufFrames, 4049 "Bad bookkeeping while updating frames pending. Timed buffer is" 4050 " only %u bytes long, but claims to have consumed %u" 4051 " bytes. (update reason: \"%s\")", 4052 bufFrames, consumedAlready, logTag); 4053 4054 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4055 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4056 "Bad bookkeeping while updating frames pending. Should have at" 4057 " least %u queued frames, but we think we have only %u. (update" 4058 " reason: \"%s\")", 4059 bufFrames, mFramesPendingInQueue, logTag); 4060 4061 mFramesPendingInQueue -= bufFrames; 4062} 4063 4064status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4065 const sp<IMemory>& buffer, int64_t pts) { 4066 4067 { 4068 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4069 if (!mMediaTimeTransformValid) 4070 return INVALID_OPERATION; 4071 } 4072 4073 Mutex::Autolock _l(mTimedBufferQueueLock); 4074 4075 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4076 mFramesPendingInQueue += bufFrames; 4077 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4078 4079 return NO_ERROR; 4080} 4081 4082status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4083 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4084 4085 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4086 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4087 target); 4088 4089 if (!(target == TimedAudioTrack::LOCAL_TIME || 4090 target == TimedAudioTrack::COMMON_TIME)) { 4091 return BAD_VALUE; 4092 } 4093 4094 Mutex::Autolock lock(mMediaTimeTransformLock); 4095 mMediaTimeTransform = xform; 4096 mMediaTimeTransformTarget = target; 4097 mMediaTimeTransformValid = true; 4098 4099 return NO_ERROR; 4100} 4101 4102#define min(a, b) ((a) < (b) ? (a) : (b)) 4103 4104// implementation of getNextBuffer for tracks whose buffers have timestamps 4105status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4106 AudioBufferProvider::Buffer* buffer, int64_t pts) 4107{ 4108 if (pts == AudioBufferProvider::kInvalidPTS) { 4109 buffer->raw = 0; 4110 buffer->frameCount = 0; 4111 return INVALID_OPERATION; 4112 } 4113 4114 Mutex::Autolock _l(mTimedBufferQueueLock); 4115 4116 ALOG_ASSERT(!mQueueHeadInFlight, 4117 "getNextBuffer called without releaseBuffer!"); 4118 4119 while (true) { 4120 4121 // if we have no timed buffers, then fail 4122 if (mTimedBufferQueue.isEmpty()) { 4123 buffer->raw = 0; 4124 buffer->frameCount = 0; 4125 return NOT_ENOUGH_DATA; 4126 } 4127 4128 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4129 4130 // calculate the PTS of the head of the timed buffer queue expressed in 4131 // local time 4132 int64_t headLocalPTS; 4133 { 4134 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4135 4136 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4137 4138 if (mMediaTimeTransform.a_to_b_denom == 0) { 4139 // the transform represents a pause, so yield silence 4140 timedYieldSilence_l(buffer->frameCount, buffer); 4141 return NO_ERROR; 4142 } 4143 4144 int64_t transformedPTS; 4145 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4146 &transformedPTS)) { 4147 // the transform failed. this shouldn't happen, but if it does 4148 // then just drop this buffer 4149 ALOGW("timedGetNextBuffer transform failed"); 4150 buffer->raw = 0; 4151 buffer->frameCount = 0; 4152 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4153 return NO_ERROR; 4154 } 4155 4156 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4157 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4158 &headLocalPTS)) { 4159 buffer->raw = 0; 4160 buffer->frameCount = 0; 4161 return INVALID_OPERATION; 4162 } 4163 } else { 4164 headLocalPTS = transformedPTS; 4165 } 4166 } 4167 4168 // adjust the head buffer's PTS to reflect the portion of the head buffer 4169 // that has already been consumed 4170 int64_t effectivePTS = headLocalPTS + 4171 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4172 4173 // Calculate the delta in samples between the head of the input buffer 4174 // queue and the start of the next output buffer that will be written. 4175 // If the transformation fails because of over or underflow, it means 4176 // that the sample's position in the output stream is so far out of 4177 // whack that it should just be dropped. 4178 int64_t sampleDelta; 4179 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4180 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4181 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4182 " mix"); 4183 continue; 4184 } 4185 if (!mLocalTimeToSampleTransform.doForwardTransform( 4186 (effectivePTS - pts) << 32, &sampleDelta)) { 4187 ALOGV("*** too late during sample rate transform: dropped buffer"); 4188 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 4189 continue; 4190 } 4191 4192 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 4193 " sampleDelta=[%d.%08x]", 4194 head.pts(), head.position(), pts, 4195 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 4196 + (sampleDelta >> 32)), 4197 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4198 4199 // if the delta between the ideal placement for the next input sample and 4200 // the current output position is within this threshold, then we will 4201 // concatenate the next input samples to the previous output 4202 const int64_t kSampleContinuityThreshold = 4203 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4204 4205 // if this is the first buffer of audio that we're emitting from this track 4206 // then it should be almost exactly on time. 4207 const int64_t kSampleStartupThreshold = 1LL << 32; 4208 4209 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4210 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4211 // the next input is close enough to being on time, so concatenate it 4212 // with the last output 4213 timedYieldSamples_l(buffer); 4214 4215 ALOGVV("*** on time: head.pos=%d frameCount=%u", 4216 head.position(), buffer->frameCount); 4217 return NO_ERROR; 4218 } else if (sampleDelta > 0) { 4219 // the gap between the current output position and the proper start of 4220 // the next input sample is too big, so fill it with silence 4221 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4222 4223 timedYieldSilence_l(framesUntilNextInput, buffer); 4224 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4225 return NO_ERROR; 4226 } else { 4227 // the next input sample is late 4228 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4229 size_t onTimeSamplePosition = 4230 head.position() + lateFrames * mCblk->frameSize; 4231 4232 if (onTimeSamplePosition > head.buffer()->size()) { 4233 // all the remaining samples in the head are too late, so 4234 // drop it and move on 4235 ALOGV("*** too late: dropped buffer"); 4236 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 4237 continue; 4238 } else { 4239 // skip over the late samples 4240 head.setPosition(onTimeSamplePosition); 4241 4242 // yield the available samples 4243 timedYieldSamples_l(buffer); 4244 4245 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4246 return NO_ERROR; 4247 } 4248 } 4249 } 4250} 4251 4252// Yield samples from the timed buffer queue head up to the given output 4253// buffer's capacity. 4254// 4255// Caller must hold mTimedBufferQueueLock 4256void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 4257 AudioBufferProvider::Buffer* buffer) { 4258 4259 const TimedBuffer& head = mTimedBufferQueue[0]; 4260 4261 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4262 head.position()); 4263 4264 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4265 mCblk->frameSize); 4266 size_t framesRequested = buffer->frameCount; 4267 buffer->frameCount = min(framesLeftInHead, framesRequested); 4268 4269 mQueueHeadInFlight = true; 4270 mTimedAudioOutputOnTime = true; 4271} 4272 4273// Yield samples of silence up to the given output buffer's capacity 4274// 4275// Caller must hold mTimedBufferQueueLock 4276void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 4277 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4278 4279 // lazily allocate a buffer filled with silence 4280 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4281 delete [] mTimedSilenceBuffer; 4282 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4283 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4284 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4285 } 4286 4287 buffer->raw = mTimedSilenceBuffer; 4288 size_t framesRequested = buffer->frameCount; 4289 buffer->frameCount = min(numFrames, framesRequested); 4290 4291 mTimedAudioOutputOnTime = false; 4292} 4293 4294// AudioBufferProvider interface 4295void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4296 AudioBufferProvider::Buffer* buffer) { 4297 4298 Mutex::Autolock _l(mTimedBufferQueueLock); 4299 4300 // If the buffer which was just released is part of the buffer at the head 4301 // of the queue, be sure to update the amt of the buffer which has been 4302 // consumed. If the buffer being returned is not part of the head of the 4303 // queue, its either because the buffer is part of the silence buffer, or 4304 // because the head of the timed queue was trimmed after the mixer called 4305 // getNextBuffer but before the mixer called releaseBuffer. 4306 if (buffer->raw == mTimedSilenceBuffer) { 4307 ALOG_ASSERT(!mQueueHeadInFlight, 4308 "Queue head in flight during release of silence buffer!"); 4309 goto done; 4310 } 4311 4312 ALOG_ASSERT(mQueueHeadInFlight, 4313 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 4314 " head in flight."); 4315 4316 if (mTimedBufferQueue.size()) { 4317 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4318 4319 void* start = head.buffer()->pointer(); 4320 void* end = reinterpret_cast<void*>( 4321 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 4322 + head.buffer()->size()); 4323 4324 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 4325 "released buffer not within the head of the timed buffer" 4326 " queue; qHead = [%p, %p], released buffer = %p", 4327 start, end, buffer->raw); 4328 4329 head.setPosition(head.position() + 4330 (buffer->frameCount * mCblk->frameSize)); 4331 mQueueHeadInFlight = false; 4332 4333 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 4334 "Bad bookkeeping during releaseBuffer! Should have at" 4335 " least %u queued frames, but we think we have only %u", 4336 buffer->frameCount, mFramesPendingInQueue); 4337 4338 mFramesPendingInQueue -= buffer->frameCount; 4339 4340 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 4341 || mTrimQueueHeadOnRelease) { 4342 trimTimedBufferQueueHead_l("releaseBuffer"); 4343 mTrimQueueHeadOnRelease = false; 4344 } 4345 } else { 4346 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 4347 " buffers in the timed buffer queue"); 4348 } 4349 4350done: 4351 buffer->raw = 0; 4352 buffer->frameCount = 0; 4353} 4354 4355uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4356 Mutex::Autolock _l(mTimedBufferQueueLock); 4357 return mFramesPendingInQueue; 4358} 4359 4360AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4361 : mPTS(0), mPosition(0) {} 4362 4363AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4364 const sp<IMemory>& buffer, int64_t pts) 4365 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4366 4367// ---------------------------------------------------------------------------- 4368 4369// RecordTrack constructor must be called with AudioFlinger::mLock held 4370AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4371 RecordThread *thread, 4372 const sp<Client>& client, 4373 uint32_t sampleRate, 4374 audio_format_t format, 4375 uint32_t channelMask, 4376 int frameCount, 4377 int sessionId) 4378 : TrackBase(thread, client, sampleRate, format, 4379 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4380 mOverflow(false) 4381{ 4382 if (mCblk != NULL) { 4383 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4384 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4385 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4386 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4387 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4388 } else { 4389 mCblk->frameSize = sizeof(int8_t); 4390 } 4391 } 4392} 4393 4394AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4395{ 4396 sp<ThreadBase> thread = mThread.promote(); 4397 if (thread != 0) { 4398 AudioSystem::releaseInput(thread->id()); 4399 } 4400} 4401 4402// AudioBufferProvider interface 4403status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4404{ 4405 audio_track_cblk_t* cblk = this->cblk(); 4406 uint32_t framesAvail; 4407 uint32_t framesReq = buffer->frameCount; 4408 4409 // Check if last stepServer failed, try to step now 4410 if (mStepServerFailed) { 4411 if (!step()) goto getNextBuffer_exit; 4412 ALOGV("stepServer recovered"); 4413 mStepServerFailed = false; 4414 } 4415 4416 framesAvail = cblk->framesAvailable_l(); 4417 4418 if (CC_LIKELY(framesAvail)) { 4419 uint32_t s = cblk->server; 4420 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4421 4422 if (framesReq > framesAvail) { 4423 framesReq = framesAvail; 4424 } 4425 if (framesReq > bufferEnd - s) { 4426 framesReq = bufferEnd - s; 4427 } 4428 4429 buffer->raw = getBuffer(s, framesReq); 4430 if (buffer->raw == NULL) goto getNextBuffer_exit; 4431 4432 buffer->frameCount = framesReq; 4433 return NO_ERROR; 4434 } 4435 4436getNextBuffer_exit: 4437 buffer->raw = NULL; 4438 buffer->frameCount = 0; 4439 return NOT_ENOUGH_DATA; 4440} 4441 4442status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid, 4443 AudioSystem::sync_event_t event, 4444 int triggerSession) 4445{ 4446 sp<ThreadBase> thread = mThread.promote(); 4447 if (thread != 0) { 4448 RecordThread *recordThread = (RecordThread *)thread.get(); 4449 return recordThread->start(this, tid, event, triggerSession); 4450 } else { 4451 return BAD_VALUE; 4452 } 4453} 4454 4455void AudioFlinger::RecordThread::RecordTrack::stop() 4456{ 4457 sp<ThreadBase> thread = mThread.promote(); 4458 if (thread != 0) { 4459 RecordThread *recordThread = (RecordThread *)thread.get(); 4460 recordThread->stop(this); 4461 TrackBase::reset(); 4462 // Force overrun condition to avoid false overrun callback until first data is 4463 // read from buffer 4464 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4465 } 4466} 4467 4468void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4469{ 4470 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4471 (mClient == 0) ? getpid_cached : mClient->pid(), 4472 mFormat, 4473 mChannelMask, 4474 mSessionId, 4475 mFrameCount, 4476 mState, 4477 mCblk->sampleRate, 4478 mCblk->server, 4479 mCblk->user); 4480} 4481 4482 4483// ---------------------------------------------------------------------------- 4484 4485AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4486 PlaybackThread *playbackThread, 4487 DuplicatingThread *sourceThread, 4488 uint32_t sampleRate, 4489 audio_format_t format, 4490 uint32_t channelMask, 4491 int frameCount) 4492 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 4493 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 4494 mActive(false), mSourceThread(sourceThread) 4495{ 4496 4497 if (mCblk != NULL) { 4498 mCblk->flags |= CBLK_DIRECTION_OUT; 4499 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4500 mOutBuffer.frameCount = 0; 4501 playbackThread->mTracks.add(this); 4502 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4503 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4504 mCblk, mBuffer, mCblk->buffers, 4505 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4506 } else { 4507 ALOGW("Error creating output track on thread %p", playbackThread); 4508 } 4509} 4510 4511AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4512{ 4513 clearBufferQueue(); 4514} 4515 4516status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid, 4517 AudioSystem::sync_event_t event, 4518 int triggerSession) 4519{ 4520 status_t status = Track::start(tid, event, triggerSession); 4521 if (status != NO_ERROR) { 4522 return status; 4523 } 4524 4525 mActive = true; 4526 mRetryCount = 127; 4527 return status; 4528} 4529 4530void AudioFlinger::PlaybackThread::OutputTrack::stop() 4531{ 4532 Track::stop(); 4533 clearBufferQueue(); 4534 mOutBuffer.frameCount = 0; 4535 mActive = false; 4536} 4537 4538bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4539{ 4540 Buffer *pInBuffer; 4541 Buffer inBuffer; 4542 uint32_t channelCount = mChannelCount; 4543 bool outputBufferFull = false; 4544 inBuffer.frameCount = frames; 4545 inBuffer.i16 = data; 4546 4547 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4548 4549 if (!mActive && frames != 0) { 4550 start(0); 4551 sp<ThreadBase> thread = mThread.promote(); 4552 if (thread != 0) { 4553 MixerThread *mixerThread = (MixerThread *)thread.get(); 4554 if (mCblk->frameCount > frames){ 4555 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4556 uint32_t startFrames = (mCblk->frameCount - frames); 4557 pInBuffer = new Buffer; 4558 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4559 pInBuffer->frameCount = startFrames; 4560 pInBuffer->i16 = pInBuffer->mBuffer; 4561 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4562 mBufferQueue.add(pInBuffer); 4563 } else { 4564 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4565 } 4566 } 4567 } 4568 } 4569 4570 while (waitTimeLeftMs) { 4571 // First write pending buffers, then new data 4572 if (mBufferQueue.size()) { 4573 pInBuffer = mBufferQueue.itemAt(0); 4574 } else { 4575 pInBuffer = &inBuffer; 4576 } 4577 4578 if (pInBuffer->frameCount == 0) { 4579 break; 4580 } 4581 4582 if (mOutBuffer.frameCount == 0) { 4583 mOutBuffer.frameCount = pInBuffer->frameCount; 4584 nsecs_t startTime = systemTime(); 4585 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4586 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4587 outputBufferFull = true; 4588 break; 4589 } 4590 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4591 if (waitTimeLeftMs >= waitTimeMs) { 4592 waitTimeLeftMs -= waitTimeMs; 4593 } else { 4594 waitTimeLeftMs = 0; 4595 } 4596 } 4597 4598 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4599 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4600 mCblk->stepUser(outFrames); 4601 pInBuffer->frameCount -= outFrames; 4602 pInBuffer->i16 += outFrames * channelCount; 4603 mOutBuffer.frameCount -= outFrames; 4604 mOutBuffer.i16 += outFrames * channelCount; 4605 4606 if (pInBuffer->frameCount == 0) { 4607 if (mBufferQueue.size()) { 4608 mBufferQueue.removeAt(0); 4609 delete [] pInBuffer->mBuffer; 4610 delete pInBuffer; 4611 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4612 } else { 4613 break; 4614 } 4615 } 4616 } 4617 4618 // If we could not write all frames, allocate a buffer and queue it for next time. 4619 if (inBuffer.frameCount) { 4620 sp<ThreadBase> thread = mThread.promote(); 4621 if (thread != 0 && !thread->standby()) { 4622 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4623 pInBuffer = new Buffer; 4624 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4625 pInBuffer->frameCount = inBuffer.frameCount; 4626 pInBuffer->i16 = pInBuffer->mBuffer; 4627 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4628 mBufferQueue.add(pInBuffer); 4629 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4630 } else { 4631 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4632 } 4633 } 4634 } 4635 4636 // Calling write() with a 0 length buffer, means that no more data will be written: 4637 // If no more buffers are pending, fill output track buffer to make sure it is started 4638 // by output mixer. 4639 if (frames == 0 && mBufferQueue.size() == 0) { 4640 if (mCblk->user < mCblk->frameCount) { 4641 frames = mCblk->frameCount - mCblk->user; 4642 pInBuffer = new Buffer; 4643 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4644 pInBuffer->frameCount = frames; 4645 pInBuffer->i16 = pInBuffer->mBuffer; 4646 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4647 mBufferQueue.add(pInBuffer); 4648 } else if (mActive) { 4649 stop(); 4650 } 4651 } 4652 4653 return outputBufferFull; 4654} 4655 4656status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4657{ 4658 int active; 4659 status_t result; 4660 audio_track_cblk_t* cblk = mCblk; 4661 uint32_t framesReq = buffer->frameCount; 4662 4663// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4664 buffer->frameCount = 0; 4665 4666 uint32_t framesAvail = cblk->framesAvailable(); 4667 4668 4669 if (framesAvail == 0) { 4670 Mutex::Autolock _l(cblk->lock); 4671 goto start_loop_here; 4672 while (framesAvail == 0) { 4673 active = mActive; 4674 if (CC_UNLIKELY(!active)) { 4675 ALOGV("Not active and NO_MORE_BUFFERS"); 4676 return NO_MORE_BUFFERS; 4677 } 4678 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4679 if (result != NO_ERROR) { 4680 return NO_MORE_BUFFERS; 4681 } 4682 // read the server count again 4683 start_loop_here: 4684 framesAvail = cblk->framesAvailable_l(); 4685 } 4686 } 4687 4688// if (framesAvail < framesReq) { 4689// return NO_MORE_BUFFERS; 4690// } 4691 4692 if (framesReq > framesAvail) { 4693 framesReq = framesAvail; 4694 } 4695 4696 uint32_t u = cblk->user; 4697 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4698 4699 if (framesReq > bufferEnd - u) { 4700 framesReq = bufferEnd - u; 4701 } 4702 4703 buffer->frameCount = framesReq; 4704 buffer->raw = (void *)cblk->buffer(u); 4705 return NO_ERROR; 4706} 4707 4708 4709void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4710{ 4711 size_t size = mBufferQueue.size(); 4712 4713 for (size_t i = 0; i < size; i++) { 4714 Buffer *pBuffer = mBufferQueue.itemAt(i); 4715 delete [] pBuffer->mBuffer; 4716 delete pBuffer; 4717 } 4718 mBufferQueue.clear(); 4719} 4720 4721// ---------------------------------------------------------------------------- 4722 4723AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4724 : RefBase(), 4725 mAudioFlinger(audioFlinger), 4726 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4727 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4728 mPid(pid), 4729 mTimedTrackCount(0) 4730{ 4731 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4732} 4733 4734// Client destructor must be called with AudioFlinger::mLock held 4735AudioFlinger::Client::~Client() 4736{ 4737 mAudioFlinger->removeClient_l(mPid); 4738} 4739 4740sp<MemoryDealer> AudioFlinger::Client::heap() const 4741{ 4742 return mMemoryDealer; 4743} 4744 4745// Reserve one of the limited slots for a timed audio track associated 4746// with this client 4747bool AudioFlinger::Client::reserveTimedTrack() 4748{ 4749 const int kMaxTimedTracksPerClient = 4; 4750 4751 Mutex::Autolock _l(mTimedTrackLock); 4752 4753 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4754 ALOGW("can not create timed track - pid %d has exceeded the limit", 4755 mPid); 4756 return false; 4757 } 4758 4759 mTimedTrackCount++; 4760 return true; 4761} 4762 4763// Release a slot for a timed audio track 4764void AudioFlinger::Client::releaseTimedTrack() 4765{ 4766 Mutex::Autolock _l(mTimedTrackLock); 4767 mTimedTrackCount--; 4768} 4769 4770// ---------------------------------------------------------------------------- 4771 4772AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4773 const sp<IAudioFlingerClient>& client, 4774 pid_t pid) 4775 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4776{ 4777} 4778 4779AudioFlinger::NotificationClient::~NotificationClient() 4780{ 4781} 4782 4783void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4784{ 4785 sp<NotificationClient> keep(this); 4786 mAudioFlinger->removeNotificationClient(mPid); 4787} 4788 4789// ---------------------------------------------------------------------------- 4790 4791AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4792 : BnAudioTrack(), 4793 mTrack(track) 4794{ 4795} 4796 4797AudioFlinger::TrackHandle::~TrackHandle() { 4798 // just stop the track on deletion, associated resources 4799 // will be freed from the main thread once all pending buffers have 4800 // been played. Unless it's not in the active track list, in which 4801 // case we free everything now... 4802 mTrack->destroy(); 4803} 4804 4805sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4806 return mTrack->getCblk(); 4807} 4808 4809status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4810 return mTrack->start(tid); 4811} 4812 4813void AudioFlinger::TrackHandle::stop() { 4814 mTrack->stop(); 4815} 4816 4817void AudioFlinger::TrackHandle::flush() { 4818 mTrack->flush(); 4819} 4820 4821void AudioFlinger::TrackHandle::mute(bool e) { 4822 mTrack->mute(e); 4823} 4824 4825void AudioFlinger::TrackHandle::pause() { 4826 mTrack->pause(); 4827} 4828 4829status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4830{ 4831 return mTrack->attachAuxEffect(EffectId); 4832} 4833 4834status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4835 sp<IMemory>* buffer) { 4836 if (!mTrack->isTimedTrack()) 4837 return INVALID_OPERATION; 4838 4839 PlaybackThread::TimedTrack* tt = 4840 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4841 return tt->allocateTimedBuffer(size, buffer); 4842} 4843 4844status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4845 int64_t pts) { 4846 if (!mTrack->isTimedTrack()) 4847 return INVALID_OPERATION; 4848 4849 PlaybackThread::TimedTrack* tt = 4850 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4851 return tt->queueTimedBuffer(buffer, pts); 4852} 4853 4854status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4855 const LinearTransform& xform, int target) { 4856 4857 if (!mTrack->isTimedTrack()) 4858 return INVALID_OPERATION; 4859 4860 PlaybackThread::TimedTrack* tt = 4861 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4862 return tt->setMediaTimeTransform( 4863 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4864} 4865 4866status_t AudioFlinger::TrackHandle::onTransact( 4867 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4868{ 4869 return BnAudioTrack::onTransact(code, data, reply, flags); 4870} 4871 4872// ---------------------------------------------------------------------------- 4873 4874sp<IAudioRecord> AudioFlinger::openRecord( 4875 pid_t pid, 4876 audio_io_handle_t input, 4877 uint32_t sampleRate, 4878 audio_format_t format, 4879 uint32_t channelMask, 4880 int frameCount, 4881 IAudioFlinger::track_flags_t flags, 4882 int *sessionId, 4883 status_t *status) 4884{ 4885 sp<RecordThread::RecordTrack> recordTrack; 4886 sp<RecordHandle> recordHandle; 4887 sp<Client> client; 4888 status_t lStatus; 4889 RecordThread *thread; 4890 size_t inFrameCount; 4891 int lSessionId; 4892 4893 // check calling permissions 4894 if (!recordingAllowed()) { 4895 lStatus = PERMISSION_DENIED; 4896 goto Exit; 4897 } 4898 4899 // add client to list 4900 { // scope for mLock 4901 Mutex::Autolock _l(mLock); 4902 thread = checkRecordThread_l(input); 4903 if (thread == NULL) { 4904 lStatus = BAD_VALUE; 4905 goto Exit; 4906 } 4907 4908 client = registerPid_l(pid); 4909 4910 // If no audio session id is provided, create one here 4911 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4912 lSessionId = *sessionId; 4913 } else { 4914 lSessionId = nextUniqueId(); 4915 if (sessionId != NULL) { 4916 *sessionId = lSessionId; 4917 } 4918 } 4919 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4920 recordTrack = thread->createRecordTrack_l(client, 4921 sampleRate, 4922 format, 4923 channelMask, 4924 frameCount, 4925 lSessionId, 4926 &lStatus); 4927 } 4928 if (lStatus != NO_ERROR) { 4929 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4930 // destructor is called by the TrackBase destructor with mLock held 4931 client.clear(); 4932 recordTrack.clear(); 4933 goto Exit; 4934 } 4935 4936 // return to handle to client 4937 recordHandle = new RecordHandle(recordTrack); 4938 lStatus = NO_ERROR; 4939 4940Exit: 4941 if (status) { 4942 *status = lStatus; 4943 } 4944 return recordHandle; 4945} 4946 4947// ---------------------------------------------------------------------------- 4948 4949AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4950 : BnAudioRecord(), 4951 mRecordTrack(recordTrack) 4952{ 4953} 4954 4955AudioFlinger::RecordHandle::~RecordHandle() { 4956 stop(); 4957} 4958 4959sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4960 return mRecordTrack->getCblk(); 4961} 4962 4963status_t AudioFlinger::RecordHandle::start(pid_t tid, int event, int triggerSession) { 4964 ALOGV("RecordHandle::start()"); 4965 return mRecordTrack->start(tid, (AudioSystem::sync_event_t)event, triggerSession); 4966} 4967 4968void AudioFlinger::RecordHandle::stop() { 4969 ALOGV("RecordHandle::stop()"); 4970 mRecordTrack->stop(); 4971} 4972 4973status_t AudioFlinger::RecordHandle::onTransact( 4974 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4975{ 4976 return BnAudioRecord::onTransact(code, data, reply, flags); 4977} 4978 4979// ---------------------------------------------------------------------------- 4980 4981AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4982 AudioStreamIn *input, 4983 uint32_t sampleRate, 4984 uint32_t channels, 4985 audio_io_handle_t id, 4986 uint32_t device) : 4987 ThreadBase(audioFlinger, id, device, RECORD), 4988 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4989 // mRsmpInIndex and mInputBytes set by readInputParameters() 4990 mReqChannelCount(popcount(channels)), 4991 mReqSampleRate(sampleRate) 4992 // mBytesRead is only meaningful while active, and so is cleared in start() 4993 // (but might be better to also clear here for dump?) 4994{ 4995 snprintf(mName, kNameLength, "AudioIn_%X", id); 4996 4997 readInputParameters(); 4998} 4999 5000 5001AudioFlinger::RecordThread::~RecordThread() 5002{ 5003 delete[] mRsmpInBuffer; 5004 delete mResampler; 5005 delete[] mRsmpOutBuffer; 5006} 5007 5008void AudioFlinger::RecordThread::onFirstRef() 5009{ 5010 run(mName, PRIORITY_URGENT_AUDIO); 5011} 5012 5013status_t AudioFlinger::RecordThread::readyToRun() 5014{ 5015 status_t status = initCheck(); 5016 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5017 return status; 5018} 5019 5020bool AudioFlinger::RecordThread::threadLoop() 5021{ 5022 AudioBufferProvider::Buffer buffer; 5023 sp<RecordTrack> activeTrack; 5024 Vector< sp<EffectChain> > effectChains; 5025 5026 nsecs_t lastWarning = 0; 5027 5028 acquireWakeLock(); 5029 5030 // start recording 5031 while (!exitPending()) { 5032 5033 processConfigEvents(); 5034 5035 { // scope for mLock 5036 Mutex::Autolock _l(mLock); 5037 checkForNewParameters_l(); 5038 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5039 if (!mStandby) { 5040 mInput->stream->common.standby(&mInput->stream->common); 5041 mStandby = true; 5042 } 5043 5044 if (exitPending()) break; 5045 5046 releaseWakeLock_l(); 5047 ALOGV("RecordThread: loop stopping"); 5048 // go to sleep 5049 mWaitWorkCV.wait(mLock); 5050 ALOGV("RecordThread: loop starting"); 5051 acquireWakeLock_l(); 5052 continue; 5053 } 5054 if (mActiveTrack != 0) { 5055 if (mActiveTrack->mState == TrackBase::PAUSING) { 5056 if (!mStandby) { 5057 mInput->stream->common.standby(&mInput->stream->common); 5058 mStandby = true; 5059 } 5060 mActiveTrack.clear(); 5061 mStartStopCond.broadcast(); 5062 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5063 if (mReqChannelCount != mActiveTrack->channelCount()) { 5064 mActiveTrack.clear(); 5065 mStartStopCond.broadcast(); 5066 } else if (mBytesRead != 0) { 5067 // record start succeeds only if first read from audio input 5068 // succeeds 5069 if (mBytesRead > 0) { 5070 mActiveTrack->mState = TrackBase::ACTIVE; 5071 } else { 5072 mActiveTrack.clear(); 5073 } 5074 mStartStopCond.broadcast(); 5075 } 5076 mStandby = false; 5077 } 5078 } 5079 lockEffectChains_l(effectChains); 5080 } 5081 5082 if (mActiveTrack != 0) { 5083 if (mActiveTrack->mState != TrackBase::ACTIVE && 5084 mActiveTrack->mState != TrackBase::RESUMING) { 5085 unlockEffectChains(effectChains); 5086 usleep(kRecordThreadSleepUs); 5087 continue; 5088 } 5089 for (size_t i = 0; i < effectChains.size(); i ++) { 5090 effectChains[i]->process_l(); 5091 } 5092 5093 buffer.frameCount = mFrameCount; 5094 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5095 size_t framesOut = buffer.frameCount; 5096 if (mResampler == NULL) { 5097 // no resampling 5098 while (framesOut) { 5099 size_t framesIn = mFrameCount - mRsmpInIndex; 5100 if (framesIn) { 5101 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5102 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5103 if (framesIn > framesOut) 5104 framesIn = framesOut; 5105 mRsmpInIndex += framesIn; 5106 framesOut -= framesIn; 5107 if ((int)mChannelCount == mReqChannelCount || 5108 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5109 memcpy(dst, src, framesIn * mFrameSize); 5110 } else { 5111 int16_t *src16 = (int16_t *)src; 5112 int16_t *dst16 = (int16_t *)dst; 5113 if (mChannelCount == 1) { 5114 while (framesIn--) { 5115 *dst16++ = *src16; 5116 *dst16++ = *src16++; 5117 } 5118 } else { 5119 while (framesIn--) { 5120 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5121 src16 += 2; 5122 } 5123 } 5124 } 5125 } 5126 if (framesOut && mFrameCount == mRsmpInIndex) { 5127 if (framesOut == mFrameCount && 5128 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5129 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5130 framesOut = 0; 5131 } else { 5132 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5133 mRsmpInIndex = 0; 5134 } 5135 if (mBytesRead < 0) { 5136 ALOGE("Error reading audio input"); 5137 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5138 // Force input into standby so that it tries to 5139 // recover at next read attempt 5140 mInput->stream->common.standby(&mInput->stream->common); 5141 usleep(kRecordThreadSleepUs); 5142 } 5143 mRsmpInIndex = mFrameCount; 5144 framesOut = 0; 5145 buffer.frameCount = 0; 5146 } 5147 } 5148 } 5149 } else { 5150 // resampling 5151 5152 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5153 // alter output frame count as if we were expecting stereo samples 5154 if (mChannelCount == 1 && mReqChannelCount == 1) { 5155 framesOut >>= 1; 5156 } 5157 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5158 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5159 // are 32 bit aligned which should be always true. 5160 if (mChannelCount == 2 && mReqChannelCount == 1) { 5161 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5162 // the resampler always outputs stereo samples: do post stereo to mono conversion 5163 int16_t *src = (int16_t *)mRsmpOutBuffer; 5164 int16_t *dst = buffer.i16; 5165 while (framesOut--) { 5166 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5167 src += 2; 5168 } 5169 } else { 5170 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5171 } 5172 5173 } 5174 if (mFramestoDrop == 0) { 5175 mActiveTrack->releaseBuffer(&buffer); 5176 } else { 5177 if (mFramestoDrop > 0) { 5178 mFramestoDrop -= buffer.frameCount; 5179 if (mFramestoDrop < 0) { 5180 mFramestoDrop = 0; 5181 } 5182 } 5183 } 5184 mActiveTrack->overflow(); 5185 } 5186 // client isn't retrieving buffers fast enough 5187 else { 5188 if (!mActiveTrack->setOverflow()) { 5189 nsecs_t now = systemTime(); 5190 if ((now - lastWarning) > kWarningThrottleNs) { 5191 ALOGW("RecordThread: buffer overflow"); 5192 lastWarning = now; 5193 } 5194 } 5195 // Release the processor for a while before asking for a new buffer. 5196 // This will give the application more chance to read from the buffer and 5197 // clear the overflow. 5198 usleep(kRecordThreadSleepUs); 5199 } 5200 } 5201 // enable changes in effect chain 5202 unlockEffectChains(effectChains); 5203 effectChains.clear(); 5204 } 5205 5206 if (!mStandby) { 5207 mInput->stream->common.standby(&mInput->stream->common); 5208 } 5209 mActiveTrack.clear(); 5210 5211 mStartStopCond.broadcast(); 5212 5213 releaseWakeLock(); 5214 5215 ALOGV("RecordThread %p exiting", this); 5216 return false; 5217} 5218 5219 5220sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5221 const sp<AudioFlinger::Client>& client, 5222 uint32_t sampleRate, 5223 audio_format_t format, 5224 int channelMask, 5225 int frameCount, 5226 int sessionId, 5227 status_t *status) 5228{ 5229 sp<RecordTrack> track; 5230 status_t lStatus; 5231 5232 lStatus = initCheck(); 5233 if (lStatus != NO_ERROR) { 5234 ALOGE("Audio driver not initialized."); 5235 goto Exit; 5236 } 5237 5238 { // scope for mLock 5239 Mutex::Autolock _l(mLock); 5240 5241 track = new RecordTrack(this, client, sampleRate, 5242 format, channelMask, frameCount, sessionId); 5243 5244 if (track->getCblk() == 0) { 5245 lStatus = NO_MEMORY; 5246 goto Exit; 5247 } 5248 5249 mTrack = track.get(); 5250 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5251 bool suspend = audio_is_bluetooth_sco_device( 5252 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5253 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5254 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5255 } 5256 lStatus = NO_ERROR; 5257 5258Exit: 5259 if (status) { 5260 *status = lStatus; 5261 } 5262 return track; 5263} 5264 5265status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5266 pid_t tid, AudioSystem::sync_event_t event, 5267 int triggerSession) 5268{ 5269 ALOGV("RecordThread::start tid=%d, event %d, triggerSession %d", tid, event, triggerSession); 5270 sp<ThreadBase> strongMe = this; 5271 status_t status = NO_ERROR; 5272 5273 if (event == AudioSystem::SYNC_EVENT_NONE) { 5274 mSyncStartEvent.clear(); 5275 mFramestoDrop = 0; 5276 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5277 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5278 triggerSession, 5279 recordTrack->sessionId(), 5280 syncStartEventCallback, 5281 this); 5282 mFramestoDrop = -1; 5283 } 5284 5285 { 5286 AutoMutex lock(mLock); 5287 if (mActiveTrack != 0) { 5288 if (recordTrack != mActiveTrack.get()) { 5289 status = -EBUSY; 5290 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5291 mActiveTrack->mState = TrackBase::ACTIVE; 5292 } 5293 return status; 5294 } 5295 5296 recordTrack->mState = TrackBase::IDLE; 5297 mActiveTrack = recordTrack; 5298 mLock.unlock(); 5299 status_t status = AudioSystem::startInput(mId); 5300 mLock.lock(); 5301 if (status != NO_ERROR) { 5302 mActiveTrack.clear(); 5303 clearSyncStartEvent(); 5304 return status; 5305 } 5306 mRsmpInIndex = mFrameCount; 5307 mBytesRead = 0; 5308 if (mResampler != NULL) { 5309 mResampler->reset(); 5310 } 5311 mActiveTrack->mState = TrackBase::RESUMING; 5312 // signal thread to start 5313 ALOGV("Signal record thread"); 5314 mWaitWorkCV.signal(); 5315 // do not wait for mStartStopCond if exiting 5316 if (exitPending()) { 5317 mActiveTrack.clear(); 5318 status = INVALID_OPERATION; 5319 goto startError; 5320 } 5321 mStartStopCond.wait(mLock); 5322 if (mActiveTrack == 0) { 5323 ALOGV("Record failed to start"); 5324 status = BAD_VALUE; 5325 goto startError; 5326 } 5327 ALOGV("Record started OK"); 5328 return status; 5329 } 5330startError: 5331 AudioSystem::stopInput(mId); 5332 clearSyncStartEvent(); 5333 return status; 5334} 5335 5336void AudioFlinger::RecordThread::clearSyncStartEvent() 5337{ 5338 if (mSyncStartEvent != 0) { 5339 mSyncStartEvent->cancel(); 5340 } 5341 mSyncStartEvent.clear(); 5342} 5343 5344void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5345{ 5346 sp<SyncEvent> strongEvent = event.promote(); 5347 5348 if (strongEvent != 0) { 5349 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5350 me->handleSyncStartEvent(strongEvent); 5351 } 5352} 5353 5354void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5355{ 5356 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5357 mActiveTrack.get(), 5358 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5359 event->listenerSession()); 5360 5361 if (mActiveTrack != 0 && 5362 event == mSyncStartEvent) { 5363 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5364 // from audio HAL 5365 mFramestoDrop = mFrameCount * 2; 5366 mSyncStartEvent.clear(); 5367 } 5368} 5369 5370void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5371 ALOGV("RecordThread::stop"); 5372 sp<ThreadBase> strongMe = this; 5373 { 5374 AutoMutex lock(mLock); 5375 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5376 mActiveTrack->mState = TrackBase::PAUSING; 5377 // do not wait for mStartStopCond if exiting 5378 if (exitPending()) { 5379 return; 5380 } 5381 mStartStopCond.wait(mLock); 5382 // if we have been restarted, recordTrack == mActiveTrack.get() here 5383 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5384 mLock.unlock(); 5385 AudioSystem::stopInput(mId); 5386 mLock.lock(); 5387 ALOGV("Record stopped OK"); 5388 } 5389 } 5390 } 5391} 5392 5393bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5394{ 5395 return false; 5396} 5397 5398status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5399{ 5400 if (!isValidSyncEvent(event)) { 5401 return BAD_VALUE; 5402 } 5403 5404 Mutex::Autolock _l(mLock); 5405 5406 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5407 mTrack->setSyncEvent(event); 5408 return NO_ERROR; 5409 } 5410 return NAME_NOT_FOUND; 5411} 5412 5413status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5414{ 5415 const size_t SIZE = 256; 5416 char buffer[SIZE]; 5417 String8 result; 5418 5419 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5420 result.append(buffer); 5421 5422 if (mActiveTrack != 0) { 5423 result.append("Active Track:\n"); 5424 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5425 mActiveTrack->dump(buffer, SIZE); 5426 result.append(buffer); 5427 5428 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5429 result.append(buffer); 5430 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5431 result.append(buffer); 5432 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5433 result.append(buffer); 5434 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5435 result.append(buffer); 5436 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5437 result.append(buffer); 5438 5439 5440 } else { 5441 result.append("No record client\n"); 5442 } 5443 write(fd, result.string(), result.size()); 5444 5445 dumpBase(fd, args); 5446 dumpEffectChains(fd, args); 5447 5448 return NO_ERROR; 5449} 5450 5451// AudioBufferProvider interface 5452status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5453{ 5454 size_t framesReq = buffer->frameCount; 5455 size_t framesReady = mFrameCount - mRsmpInIndex; 5456 int channelCount; 5457 5458 if (framesReady == 0) { 5459 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5460 if (mBytesRead < 0) { 5461 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5462 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5463 // Force input into standby so that it tries to 5464 // recover at next read attempt 5465 mInput->stream->common.standby(&mInput->stream->common); 5466 usleep(kRecordThreadSleepUs); 5467 } 5468 buffer->raw = NULL; 5469 buffer->frameCount = 0; 5470 return NOT_ENOUGH_DATA; 5471 } 5472 mRsmpInIndex = 0; 5473 framesReady = mFrameCount; 5474 } 5475 5476 if (framesReq > framesReady) { 5477 framesReq = framesReady; 5478 } 5479 5480 if (mChannelCount == 1 && mReqChannelCount == 2) { 5481 channelCount = 1; 5482 } else { 5483 channelCount = 2; 5484 } 5485 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5486 buffer->frameCount = framesReq; 5487 return NO_ERROR; 5488} 5489 5490// AudioBufferProvider interface 5491void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5492{ 5493 mRsmpInIndex += buffer->frameCount; 5494 buffer->frameCount = 0; 5495} 5496 5497bool AudioFlinger::RecordThread::checkForNewParameters_l() 5498{ 5499 bool reconfig = false; 5500 5501 while (!mNewParameters.isEmpty()) { 5502 status_t status = NO_ERROR; 5503 String8 keyValuePair = mNewParameters[0]; 5504 AudioParameter param = AudioParameter(keyValuePair); 5505 int value; 5506 audio_format_t reqFormat = mFormat; 5507 int reqSamplingRate = mReqSampleRate; 5508 int reqChannelCount = mReqChannelCount; 5509 5510 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5511 reqSamplingRate = value; 5512 reconfig = true; 5513 } 5514 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5515 reqFormat = (audio_format_t) value; 5516 reconfig = true; 5517 } 5518 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5519 reqChannelCount = popcount(value); 5520 reconfig = true; 5521 } 5522 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5523 // do not accept frame count changes if tracks are open as the track buffer 5524 // size depends on frame count and correct behavior would not be guaranteed 5525 // if frame count is changed after track creation 5526 if (mActiveTrack != 0) { 5527 status = INVALID_OPERATION; 5528 } else { 5529 reconfig = true; 5530 } 5531 } 5532 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5533 // forward device change to effects that have requested to be 5534 // aware of attached audio device. 5535 for (size_t i = 0; i < mEffectChains.size(); i++) { 5536 mEffectChains[i]->setDevice_l(value); 5537 } 5538 // store input device and output device but do not forward output device to audio HAL. 5539 // Note that status is ignored by the caller for output device 5540 // (see AudioFlinger::setParameters() 5541 if (value & AUDIO_DEVICE_OUT_ALL) { 5542 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5543 status = BAD_VALUE; 5544 } else { 5545 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5546 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5547 if (mTrack != NULL) { 5548 bool suspend = audio_is_bluetooth_sco_device( 5549 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5550 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5551 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5552 } 5553 } 5554 mDevice |= (uint32_t)value; 5555 } 5556 if (status == NO_ERROR) { 5557 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5558 if (status == INVALID_OPERATION) { 5559 mInput->stream->common.standby(&mInput->stream->common); 5560 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5561 keyValuePair.string()); 5562 } 5563 if (reconfig) { 5564 if (status == BAD_VALUE && 5565 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5566 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5567 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5568 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5569 (reqChannelCount <= FCC_2)) { 5570 status = NO_ERROR; 5571 } 5572 if (status == NO_ERROR) { 5573 readInputParameters(); 5574 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5575 } 5576 } 5577 } 5578 5579 mNewParameters.removeAt(0); 5580 5581 mParamStatus = status; 5582 mParamCond.signal(); 5583 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5584 // already timed out waiting for the status and will never signal the condition. 5585 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5586 } 5587 return reconfig; 5588} 5589 5590String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5591{ 5592 char *s; 5593 String8 out_s8 = String8(); 5594 5595 Mutex::Autolock _l(mLock); 5596 if (initCheck() != NO_ERROR) { 5597 return out_s8; 5598 } 5599 5600 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5601 out_s8 = String8(s); 5602 free(s); 5603 return out_s8; 5604} 5605 5606void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5607 AudioSystem::OutputDescriptor desc; 5608 void *param2 = NULL; 5609 5610 switch (event) { 5611 case AudioSystem::INPUT_OPENED: 5612 case AudioSystem::INPUT_CONFIG_CHANGED: 5613 desc.channels = mChannelMask; 5614 desc.samplingRate = mSampleRate; 5615 desc.format = mFormat; 5616 desc.frameCount = mFrameCount; 5617 desc.latency = 0; 5618 param2 = &desc; 5619 break; 5620 5621 case AudioSystem::INPUT_CLOSED: 5622 default: 5623 break; 5624 } 5625 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5626} 5627 5628void AudioFlinger::RecordThread::readInputParameters() 5629{ 5630 delete mRsmpInBuffer; 5631 // mRsmpInBuffer is always assigned a new[] below 5632 delete mRsmpOutBuffer; 5633 mRsmpOutBuffer = NULL; 5634 delete mResampler; 5635 mResampler = NULL; 5636 5637 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5638 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5639 mChannelCount = (uint16_t)popcount(mChannelMask); 5640 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5641 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5642 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5643 mFrameCount = mInputBytes / mFrameSize; 5644 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5645 5646 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5647 { 5648 int channelCount; 5649 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5650 // stereo to mono post process as the resampler always outputs stereo. 5651 if (mChannelCount == 1 && mReqChannelCount == 2) { 5652 channelCount = 1; 5653 } else { 5654 channelCount = 2; 5655 } 5656 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5657 mResampler->setSampleRate(mSampleRate); 5658 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5659 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5660 5661 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5662 if (mChannelCount == 1 && mReqChannelCount == 1) { 5663 mFrameCount >>= 1; 5664 } 5665 5666 } 5667 mRsmpInIndex = mFrameCount; 5668} 5669 5670unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5671{ 5672 Mutex::Autolock _l(mLock); 5673 if (initCheck() != NO_ERROR) { 5674 return 0; 5675 } 5676 5677 return mInput->stream->get_input_frames_lost(mInput->stream); 5678} 5679 5680uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5681{ 5682 Mutex::Autolock _l(mLock); 5683 uint32_t result = 0; 5684 if (getEffectChain_l(sessionId) != 0) { 5685 result = EFFECT_SESSION; 5686 } 5687 5688 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5689 result |= TRACK_SESSION; 5690 } 5691 5692 return result; 5693} 5694 5695AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5696{ 5697 Mutex::Autolock _l(mLock); 5698 return mTrack; 5699} 5700 5701AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5702{ 5703 Mutex::Autolock _l(mLock); 5704 return mInput; 5705} 5706 5707AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5708{ 5709 Mutex::Autolock _l(mLock); 5710 AudioStreamIn *input = mInput; 5711 mInput = NULL; 5712 return input; 5713} 5714 5715// this method must always be called either with ThreadBase mLock held or inside the thread loop 5716audio_stream_t* AudioFlinger::RecordThread::stream() const 5717{ 5718 if (mInput == NULL) { 5719 return NULL; 5720 } 5721 return &mInput->stream->common; 5722} 5723 5724 5725// ---------------------------------------------------------------------------- 5726 5727audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 5728{ 5729 if (!settingsAllowed()) { 5730 return 0; 5731 } 5732 Mutex::Autolock _l(mLock); 5733 return loadHwModule_l(name); 5734} 5735 5736// loadHwModule_l() must be called with AudioFlinger::mLock held 5737audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 5738{ 5739 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5740 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 5741 ALOGW("loadHwModule() module %s already loaded", name); 5742 return mAudioHwDevs.keyAt(i); 5743 } 5744 } 5745 5746 const hw_module_t *mod; 5747 audio_hw_device_t *dev; 5748 5749 int rc = load_audio_interface(name, &mod, &dev); 5750 if (rc) { 5751 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 5752 return 0; 5753 } 5754 5755 mHardwareStatus = AUDIO_HW_INIT; 5756 rc = dev->init_check(dev); 5757 mHardwareStatus = AUDIO_HW_IDLE; 5758 if (rc) { 5759 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 5760 return 0; 5761 } 5762 5763 if ((mMasterVolumeSupportLvl != MVS_NONE) && 5764 (NULL != dev->set_master_volume)) { 5765 AutoMutex lock(mHardwareLock); 5766 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5767 dev->set_master_volume(dev, mMasterVolume); 5768 mHardwareStatus = AUDIO_HW_IDLE; 5769 } 5770 5771 audio_module_handle_t handle = nextUniqueId(); 5772 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 5773 5774 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 5775 name, mod->name, mod->id, handle); 5776 5777 return handle; 5778 5779} 5780 5781audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 5782 audio_devices_t *pDevices, 5783 uint32_t *pSamplingRate, 5784 audio_format_t *pFormat, 5785 audio_channel_mask_t *pChannelMask, 5786 uint32_t *pLatencyMs, 5787 audio_policy_output_flags_t flags) 5788{ 5789 status_t status; 5790 PlaybackThread *thread = NULL; 5791 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5792 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5793 audio_channel_mask_t channelMask = pChannelMask ? *pChannelMask : 0; 5794 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5795 audio_stream_out_t *outStream; 5796 audio_hw_device_t *outHwDev; 5797 5798 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5799 module, 5800 (pDevices != NULL) ? (int)*pDevices : 0, 5801 samplingRate, 5802 format, 5803 channelMask, 5804 flags); 5805 5806 if (pDevices == NULL || *pDevices == 0) { 5807 return 0; 5808 } 5809 5810 Mutex::Autolock _l(mLock); 5811 5812 outHwDev = findSuitableHwDev_l(module, *pDevices); 5813 if (outHwDev == NULL) 5814 return 0; 5815 5816 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5817 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5818 &channelMask, &samplingRate, &outStream); 5819 mHardwareStatus = AUDIO_HW_IDLE; 5820 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5821 outStream, 5822 samplingRate, 5823 format, 5824 channelMask, 5825 status); 5826 5827 if (outStream != NULL) { 5828 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5829 audio_io_handle_t id = nextUniqueId(); 5830 5831 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5832 (format != AUDIO_FORMAT_PCM_16_BIT) || 5833 (channelMask != AUDIO_CHANNEL_OUT_STEREO)) { 5834 thread = new DirectOutputThread(this, output, id, *pDevices); 5835 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5836 } else { 5837 thread = new MixerThread(this, output, id, *pDevices); 5838 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5839 } 5840 mPlaybackThreads.add(id, thread); 5841 5842 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5843 if (pFormat != NULL) *pFormat = format; 5844 if (pChannelMask != NULL) *pChannelMask = channelMask; 5845 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5846 5847 // notify client processes of the new output creation 5848 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5849 5850 // the first primary output opened designates the primary hw device 5851 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_POLICY_OUTPUT_FLAG_PRIMARY)) { 5852 ALOGI("Using module %d has the primary audio interface", module); 5853 mPrimaryHardwareDev = outHwDev; 5854 5855 AutoMutex lock(mHardwareLock); 5856 mHardwareStatus = AUDIO_HW_SET_MODE; 5857 outHwDev->set_mode(outHwDev, mMode); 5858 5859 // Determine the level of master volume support the primary audio HAL has, 5860 // and set the initial master volume at the same time. 5861 float initialVolume = 1.0; 5862 mMasterVolumeSupportLvl = MVS_NONE; 5863 5864 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 5865 if ((NULL != outHwDev->get_master_volume) && 5866 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 5867 mMasterVolumeSupportLvl = MVS_FULL; 5868 } else { 5869 mMasterVolumeSupportLvl = MVS_SETONLY; 5870 initialVolume = 1.0; 5871 } 5872 5873 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5874 if ((NULL == outHwDev->set_master_volume) || 5875 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 5876 mMasterVolumeSupportLvl = MVS_NONE; 5877 } 5878 // now that we have a primary device, initialize master volume on other devices 5879 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5880 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 5881 5882 if ((dev != mPrimaryHardwareDev) && 5883 (NULL != dev->set_master_volume)) { 5884 dev->set_master_volume(dev, initialVolume); 5885 } 5886 } 5887 mHardwareStatus = AUDIO_HW_IDLE; 5888 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 5889 ? initialVolume 5890 : 1.0; 5891 mMasterVolume = initialVolume; 5892 } 5893 return id; 5894 } 5895 5896 return 0; 5897} 5898 5899audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5900 audio_io_handle_t output2) 5901{ 5902 Mutex::Autolock _l(mLock); 5903 MixerThread *thread1 = checkMixerThread_l(output1); 5904 MixerThread *thread2 = checkMixerThread_l(output2); 5905 5906 if (thread1 == NULL || thread2 == NULL) { 5907 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5908 return 0; 5909 } 5910 5911 audio_io_handle_t id = nextUniqueId(); 5912 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5913 thread->addOutputTrack(thread2); 5914 mPlaybackThreads.add(id, thread); 5915 // notify client processes of the new output creation 5916 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5917 return id; 5918} 5919 5920status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5921{ 5922 // keep strong reference on the playback thread so that 5923 // it is not destroyed while exit() is executed 5924 sp<PlaybackThread> thread; 5925 { 5926 Mutex::Autolock _l(mLock); 5927 thread = checkPlaybackThread_l(output); 5928 if (thread == NULL) { 5929 return BAD_VALUE; 5930 } 5931 5932 ALOGV("closeOutput() %d", output); 5933 5934 if (thread->type() == ThreadBase::MIXER) { 5935 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5936 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5937 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5938 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5939 } 5940 } 5941 } 5942 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5943 mPlaybackThreads.removeItem(output); 5944 } 5945 thread->exit(); 5946 // The thread entity (active unit of execution) is no longer running here, 5947 // but the ThreadBase container still exists. 5948 5949 if (thread->type() != ThreadBase::DUPLICATING) { 5950 AudioStreamOut *out = thread->clearOutput(); 5951 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5952 // from now on thread->mOutput is NULL 5953 out->hwDev->close_output_stream(out->hwDev, out->stream); 5954 delete out; 5955 } 5956 return NO_ERROR; 5957} 5958 5959status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5960{ 5961 Mutex::Autolock _l(mLock); 5962 PlaybackThread *thread = checkPlaybackThread_l(output); 5963 5964 if (thread == NULL) { 5965 return BAD_VALUE; 5966 } 5967 5968 ALOGV("suspendOutput() %d", output); 5969 thread->suspend(); 5970 5971 return NO_ERROR; 5972} 5973 5974status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5975{ 5976 Mutex::Autolock _l(mLock); 5977 PlaybackThread *thread = checkPlaybackThread_l(output); 5978 5979 if (thread == NULL) { 5980 return BAD_VALUE; 5981 } 5982 5983 ALOGV("restoreOutput() %d", output); 5984 5985 thread->restore(); 5986 5987 return NO_ERROR; 5988} 5989 5990audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 5991 audio_devices_t *pDevices, 5992 uint32_t *pSamplingRate, 5993 audio_format_t *pFormat, 5994 uint32_t *pChannelMask) 5995{ 5996 status_t status; 5997 RecordThread *thread = NULL; 5998 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5999 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 6000 audio_channel_mask_t channelMask = pChannelMask ? *pChannelMask : 0; 6001 uint32_t reqSamplingRate = samplingRate; 6002 audio_format_t reqFormat = format; 6003 audio_channel_mask_t reqChannels = channelMask; 6004 audio_stream_in_t *inStream; 6005 audio_hw_device_t *inHwDev; 6006 6007 if (pDevices == NULL || *pDevices == 0) { 6008 return 0; 6009 } 6010 6011 Mutex::Autolock _l(mLock); 6012 6013 inHwDev = findSuitableHwDev_l(module, *pDevices); 6014 if (inHwDev == NULL) 6015 return 0; 6016 6017 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 6018 &channelMask, &samplingRate, 6019 (audio_in_acoustics_t)0, 6020 &inStream); 6021 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6022 inStream, 6023 samplingRate, 6024 format, 6025 channelMask, 6026 status); 6027 6028 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6029 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6030 // or stereo to mono conversions on 16 bit PCM inputs. 6031 if (inStream == NULL && status == BAD_VALUE && 6032 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 6033 (samplingRate <= 2 * reqSamplingRate) && 6034 (popcount(channelMask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6035 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6036 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 6037 &channelMask, &samplingRate, 6038 (audio_in_acoustics_t)0, 6039 &inStream); 6040 } 6041 6042 if (inStream != NULL) { 6043 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6044 6045 audio_io_handle_t id = nextUniqueId(); 6046 // Start record thread 6047 // RecorThread require both input and output device indication to forward to audio 6048 // pre processing modules 6049 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6050 thread = new RecordThread(this, 6051 input, 6052 reqSamplingRate, 6053 reqChannels, 6054 id, 6055 device); 6056 mRecordThreads.add(id, thread); 6057 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6058 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6059 if (pFormat != NULL) *pFormat = format; 6060 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6061 6062 input->stream->common.standby(&input->stream->common); 6063 6064 // notify client processes of the new input creation 6065 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6066 return id; 6067 } 6068 6069 return 0; 6070} 6071 6072status_t AudioFlinger::closeInput(audio_io_handle_t input) 6073{ 6074 // keep strong reference on the record thread so that 6075 // it is not destroyed while exit() is executed 6076 sp<RecordThread> thread; 6077 { 6078 Mutex::Autolock _l(mLock); 6079 thread = checkRecordThread_l(input); 6080 if (thread == NULL) { 6081 return BAD_VALUE; 6082 } 6083 6084 ALOGV("closeInput() %d", input); 6085 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6086 mRecordThreads.removeItem(input); 6087 } 6088 thread->exit(); 6089 // The thread entity (active unit of execution) is no longer running here, 6090 // but the ThreadBase container still exists. 6091 6092 AudioStreamIn *in = thread->clearInput(); 6093 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6094 // from now on thread->mInput is NULL 6095 in->hwDev->close_input_stream(in->hwDev, in->stream); 6096 delete in; 6097 6098 return NO_ERROR; 6099} 6100 6101status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6102{ 6103 Mutex::Autolock _l(mLock); 6104 MixerThread *dstThread = checkMixerThread_l(output); 6105 if (dstThread == NULL) { 6106 ALOGW("setStreamOutput() bad output id %d", output); 6107 return BAD_VALUE; 6108 } 6109 6110 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6111 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6112 6113 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6114 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6115 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6116 MixerThread *srcThread = (MixerThread *)thread; 6117 srcThread->invalidateTracks(stream); 6118 } 6119 } 6120 6121 return NO_ERROR; 6122} 6123 6124 6125int AudioFlinger::newAudioSessionId() 6126{ 6127 return nextUniqueId(); 6128} 6129 6130void AudioFlinger::acquireAudioSessionId(int audioSession) 6131{ 6132 Mutex::Autolock _l(mLock); 6133 pid_t caller = IPCThreadState::self()->getCallingPid(); 6134 ALOGV("acquiring %d from %d", audioSession, caller); 6135 size_t num = mAudioSessionRefs.size(); 6136 for (size_t i = 0; i< num; i++) { 6137 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6138 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6139 ref->mCnt++; 6140 ALOGV(" incremented refcount to %d", ref->mCnt); 6141 return; 6142 } 6143 } 6144 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6145 ALOGV(" added new entry for %d", audioSession); 6146} 6147 6148void AudioFlinger::releaseAudioSessionId(int audioSession) 6149{ 6150 Mutex::Autolock _l(mLock); 6151 pid_t caller = IPCThreadState::self()->getCallingPid(); 6152 ALOGV("releasing %d from %d", audioSession, caller); 6153 size_t num = mAudioSessionRefs.size(); 6154 for (size_t i = 0; i< num; i++) { 6155 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6156 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6157 ref->mCnt--; 6158 ALOGV(" decremented refcount to %d", ref->mCnt); 6159 if (ref->mCnt == 0) { 6160 mAudioSessionRefs.removeAt(i); 6161 delete ref; 6162 purgeStaleEffects_l(); 6163 } 6164 return; 6165 } 6166 } 6167 ALOGW("session id %d not found for pid %d", audioSession, caller); 6168} 6169 6170void AudioFlinger::purgeStaleEffects_l() { 6171 6172 ALOGV("purging stale effects"); 6173 6174 Vector< sp<EffectChain> > chains; 6175 6176 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6177 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6178 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6179 sp<EffectChain> ec = t->mEffectChains[j]; 6180 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6181 chains.push(ec); 6182 } 6183 } 6184 } 6185 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6186 sp<RecordThread> t = mRecordThreads.valueAt(i); 6187 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6188 sp<EffectChain> ec = t->mEffectChains[j]; 6189 chains.push(ec); 6190 } 6191 } 6192 6193 for (size_t i = 0; i < chains.size(); i++) { 6194 sp<EffectChain> ec = chains[i]; 6195 int sessionid = ec->sessionId(); 6196 sp<ThreadBase> t = ec->mThread.promote(); 6197 if (t == 0) { 6198 continue; 6199 } 6200 size_t numsessionrefs = mAudioSessionRefs.size(); 6201 bool found = false; 6202 for (size_t k = 0; k < numsessionrefs; k++) { 6203 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6204 if (ref->mSessionid == sessionid) { 6205 ALOGV(" session %d still exists for %d with %d refs", 6206 sessionid, ref->mPid, ref->mCnt); 6207 found = true; 6208 break; 6209 } 6210 } 6211 if (!found) { 6212 // remove all effects from the chain 6213 while (ec->mEffects.size()) { 6214 sp<EffectModule> effect = ec->mEffects[0]; 6215 effect->unPin(); 6216 Mutex::Autolock _l (t->mLock); 6217 t->removeEffect_l(effect); 6218 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6219 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6220 if (handle != 0) { 6221 handle->mEffect.clear(); 6222 if (handle->mHasControl && handle->mEnabled) { 6223 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6224 } 6225 } 6226 } 6227 AudioSystem::unregisterEffect(effect->id()); 6228 } 6229 } 6230 } 6231 return; 6232} 6233 6234// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6235AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6236{ 6237 return mPlaybackThreads.valueFor(output).get(); 6238} 6239 6240// checkMixerThread_l() must be called with AudioFlinger::mLock held 6241AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6242{ 6243 PlaybackThread *thread = checkPlaybackThread_l(output); 6244 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6245} 6246 6247// checkRecordThread_l() must be called with AudioFlinger::mLock held 6248AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6249{ 6250 return mRecordThreads.valueFor(input).get(); 6251} 6252 6253uint32_t AudioFlinger::nextUniqueId() 6254{ 6255 return android_atomic_inc(&mNextUniqueId); 6256} 6257 6258AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6259{ 6260 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6261 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6262 AudioStreamOut *output = thread->getOutput(); 6263 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6264 return thread; 6265 } 6266 } 6267 return NULL; 6268} 6269 6270uint32_t AudioFlinger::primaryOutputDevice_l() const 6271{ 6272 PlaybackThread *thread = primaryPlaybackThread_l(); 6273 6274 if (thread == NULL) { 6275 return 0; 6276 } 6277 6278 return thread->device(); 6279} 6280 6281sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6282 int triggerSession, 6283 int listenerSession, 6284 sync_event_callback_t callBack, 6285 void *cookie) 6286{ 6287 Mutex::Autolock _l(mLock); 6288 6289 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6290 status_t playStatus = NAME_NOT_FOUND; 6291 status_t recStatus = NAME_NOT_FOUND; 6292 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6293 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6294 if (playStatus == NO_ERROR) { 6295 return event; 6296 } 6297 } 6298 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6299 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6300 if (recStatus == NO_ERROR) { 6301 return event; 6302 } 6303 } 6304 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6305 mPendingSyncEvents.add(event); 6306 } else { 6307 ALOGV("createSyncEvent() invalid event %d", event->type()); 6308 event.clear(); 6309 } 6310 return event; 6311} 6312 6313// ---------------------------------------------------------------------------- 6314// Effect management 6315// ---------------------------------------------------------------------------- 6316 6317 6318status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6319{ 6320 Mutex::Autolock _l(mLock); 6321 return EffectQueryNumberEffects(numEffects); 6322} 6323 6324status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6325{ 6326 Mutex::Autolock _l(mLock); 6327 return EffectQueryEffect(index, descriptor); 6328} 6329 6330status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6331 effect_descriptor_t *descriptor) const 6332{ 6333 Mutex::Autolock _l(mLock); 6334 return EffectGetDescriptor(pUuid, descriptor); 6335} 6336 6337 6338sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6339 effect_descriptor_t *pDesc, 6340 const sp<IEffectClient>& effectClient, 6341 int32_t priority, 6342 audio_io_handle_t io, 6343 int sessionId, 6344 status_t *status, 6345 int *id, 6346 int *enabled) 6347{ 6348 status_t lStatus = NO_ERROR; 6349 sp<EffectHandle> handle; 6350 effect_descriptor_t desc; 6351 6352 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6353 pid, effectClient.get(), priority, sessionId, io); 6354 6355 if (pDesc == NULL) { 6356 lStatus = BAD_VALUE; 6357 goto Exit; 6358 } 6359 6360 // check audio settings permission for global effects 6361 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6362 lStatus = PERMISSION_DENIED; 6363 goto Exit; 6364 } 6365 6366 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6367 // that can only be created by audio policy manager (running in same process) 6368 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6369 lStatus = PERMISSION_DENIED; 6370 goto Exit; 6371 } 6372 6373 if (io == 0) { 6374 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6375 // output must be specified by AudioPolicyManager when using session 6376 // AUDIO_SESSION_OUTPUT_STAGE 6377 lStatus = BAD_VALUE; 6378 goto Exit; 6379 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6380 // if the output returned by getOutputForEffect() is removed before we lock the 6381 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6382 // and we will exit safely 6383 io = AudioSystem::getOutputForEffect(&desc); 6384 } 6385 } 6386 6387 { 6388 Mutex::Autolock _l(mLock); 6389 6390 6391 if (!EffectIsNullUuid(&pDesc->uuid)) { 6392 // if uuid is specified, request effect descriptor 6393 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6394 if (lStatus < 0) { 6395 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6396 goto Exit; 6397 } 6398 } else { 6399 // if uuid is not specified, look for an available implementation 6400 // of the required type in effect factory 6401 if (EffectIsNullUuid(&pDesc->type)) { 6402 ALOGW("createEffect() no effect type"); 6403 lStatus = BAD_VALUE; 6404 goto Exit; 6405 } 6406 uint32_t numEffects = 0; 6407 effect_descriptor_t d; 6408 d.flags = 0; // prevent compiler warning 6409 bool found = false; 6410 6411 lStatus = EffectQueryNumberEffects(&numEffects); 6412 if (lStatus < 0) { 6413 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6414 goto Exit; 6415 } 6416 for (uint32_t i = 0; i < numEffects; i++) { 6417 lStatus = EffectQueryEffect(i, &desc); 6418 if (lStatus < 0) { 6419 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6420 continue; 6421 } 6422 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6423 // If matching type found save effect descriptor. If the session is 6424 // 0 and the effect is not auxiliary, continue enumeration in case 6425 // an auxiliary version of this effect type is available 6426 found = true; 6427 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6428 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6429 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6430 break; 6431 } 6432 } 6433 } 6434 if (!found) { 6435 lStatus = BAD_VALUE; 6436 ALOGW("createEffect() effect not found"); 6437 goto Exit; 6438 } 6439 // For same effect type, chose auxiliary version over insert version if 6440 // connect to output mix (Compliance to OpenSL ES) 6441 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6442 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6443 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6444 } 6445 } 6446 6447 // Do not allow auxiliary effects on a session different from 0 (output mix) 6448 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6449 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6450 lStatus = INVALID_OPERATION; 6451 goto Exit; 6452 } 6453 6454 // check recording permission for visualizer 6455 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6456 !recordingAllowed()) { 6457 lStatus = PERMISSION_DENIED; 6458 goto Exit; 6459 } 6460 6461 // return effect descriptor 6462 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6463 6464 // If output is not specified try to find a matching audio session ID in one of the 6465 // output threads. 6466 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6467 // because of code checking output when entering the function. 6468 // Note: io is never 0 when creating an effect on an input 6469 if (io == 0) { 6470 // look for the thread where the specified audio session is present 6471 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6472 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6473 io = mPlaybackThreads.keyAt(i); 6474 break; 6475 } 6476 } 6477 if (io == 0) { 6478 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6479 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6480 io = mRecordThreads.keyAt(i); 6481 break; 6482 } 6483 } 6484 } 6485 // If no output thread contains the requested session ID, default to 6486 // first output. The effect chain will be moved to the correct output 6487 // thread when a track with the same session ID is created 6488 if (io == 0 && mPlaybackThreads.size()) { 6489 io = mPlaybackThreads.keyAt(0); 6490 } 6491 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6492 } 6493 ThreadBase *thread = checkRecordThread_l(io); 6494 if (thread == NULL) { 6495 thread = checkPlaybackThread_l(io); 6496 if (thread == NULL) { 6497 ALOGE("createEffect() unknown output thread"); 6498 lStatus = BAD_VALUE; 6499 goto Exit; 6500 } 6501 } 6502 6503 sp<Client> client = registerPid_l(pid); 6504 6505 // create effect on selected output thread 6506 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6507 &desc, enabled, &lStatus); 6508 if (handle != 0 && id != NULL) { 6509 *id = handle->id(); 6510 } 6511 } 6512 6513Exit: 6514 if (status != NULL) { 6515 *status = lStatus; 6516 } 6517 return handle; 6518} 6519 6520status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6521 audio_io_handle_t dstOutput) 6522{ 6523 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6524 sessionId, srcOutput, dstOutput); 6525 Mutex::Autolock _l(mLock); 6526 if (srcOutput == dstOutput) { 6527 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6528 return NO_ERROR; 6529 } 6530 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6531 if (srcThread == NULL) { 6532 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6533 return BAD_VALUE; 6534 } 6535 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6536 if (dstThread == NULL) { 6537 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6538 return BAD_VALUE; 6539 } 6540 6541 Mutex::Autolock _dl(dstThread->mLock); 6542 Mutex::Autolock _sl(srcThread->mLock); 6543 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6544 6545 return NO_ERROR; 6546} 6547 6548// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6549status_t AudioFlinger::moveEffectChain_l(int sessionId, 6550 AudioFlinger::PlaybackThread *srcThread, 6551 AudioFlinger::PlaybackThread *dstThread, 6552 bool reRegister) 6553{ 6554 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6555 sessionId, srcThread, dstThread); 6556 6557 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6558 if (chain == 0) { 6559 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6560 sessionId, srcThread); 6561 return INVALID_OPERATION; 6562 } 6563 6564 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6565 // so that a new chain is created with correct parameters when first effect is added. This is 6566 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6567 // removed. 6568 srcThread->removeEffectChain_l(chain); 6569 6570 // transfer all effects one by one so that new effect chain is created on new thread with 6571 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6572 audio_io_handle_t dstOutput = dstThread->id(); 6573 sp<EffectChain> dstChain; 6574 uint32_t strategy = 0; // prevent compiler warning 6575 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6576 while (effect != 0) { 6577 srcThread->removeEffect_l(effect); 6578 dstThread->addEffect_l(effect); 6579 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6580 if (effect->state() == EffectModule::ACTIVE || 6581 effect->state() == EffectModule::STOPPING) { 6582 effect->start(); 6583 } 6584 // if the move request is not received from audio policy manager, the effect must be 6585 // re-registered with the new strategy and output 6586 if (dstChain == 0) { 6587 dstChain = effect->chain().promote(); 6588 if (dstChain == 0) { 6589 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6590 srcThread->addEffect_l(effect); 6591 return NO_INIT; 6592 } 6593 strategy = dstChain->strategy(); 6594 } 6595 if (reRegister) { 6596 AudioSystem::unregisterEffect(effect->id()); 6597 AudioSystem::registerEffect(&effect->desc(), 6598 dstOutput, 6599 strategy, 6600 sessionId, 6601 effect->id()); 6602 } 6603 effect = chain->getEffectFromId_l(0); 6604 } 6605 6606 return NO_ERROR; 6607} 6608 6609 6610// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6611sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6612 const sp<AudioFlinger::Client>& client, 6613 const sp<IEffectClient>& effectClient, 6614 int32_t priority, 6615 int sessionId, 6616 effect_descriptor_t *desc, 6617 int *enabled, 6618 status_t *status 6619 ) 6620{ 6621 sp<EffectModule> effect; 6622 sp<EffectHandle> handle; 6623 status_t lStatus; 6624 sp<EffectChain> chain; 6625 bool chainCreated = false; 6626 bool effectCreated = false; 6627 bool effectRegistered = false; 6628 6629 lStatus = initCheck(); 6630 if (lStatus != NO_ERROR) { 6631 ALOGW("createEffect_l() Audio driver not initialized."); 6632 goto Exit; 6633 } 6634 6635 // Do not allow effects with session ID 0 on direct output or duplicating threads 6636 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6637 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6638 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6639 desc->name, sessionId); 6640 lStatus = BAD_VALUE; 6641 goto Exit; 6642 } 6643 // Only Pre processor effects are allowed on input threads and only on input threads 6644 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6645 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6646 desc->name, desc->flags, mType); 6647 lStatus = BAD_VALUE; 6648 goto Exit; 6649 } 6650 6651 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6652 6653 { // scope for mLock 6654 Mutex::Autolock _l(mLock); 6655 6656 // check for existing effect chain with the requested audio session 6657 chain = getEffectChain_l(sessionId); 6658 if (chain == 0) { 6659 // create a new chain for this session 6660 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6661 chain = new EffectChain(this, sessionId); 6662 addEffectChain_l(chain); 6663 chain->setStrategy(getStrategyForSession_l(sessionId)); 6664 chainCreated = true; 6665 } else { 6666 effect = chain->getEffectFromDesc_l(desc); 6667 } 6668 6669 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6670 6671 if (effect == 0) { 6672 int id = mAudioFlinger->nextUniqueId(); 6673 // Check CPU and memory usage 6674 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6675 if (lStatus != NO_ERROR) { 6676 goto Exit; 6677 } 6678 effectRegistered = true; 6679 // create a new effect module if none present in the chain 6680 effect = new EffectModule(this, chain, desc, id, sessionId); 6681 lStatus = effect->status(); 6682 if (lStatus != NO_ERROR) { 6683 goto Exit; 6684 } 6685 lStatus = chain->addEffect_l(effect); 6686 if (lStatus != NO_ERROR) { 6687 goto Exit; 6688 } 6689 effectCreated = true; 6690 6691 effect->setDevice(mDevice); 6692 effect->setMode(mAudioFlinger->getMode()); 6693 } 6694 // create effect handle and connect it to effect module 6695 handle = new EffectHandle(effect, client, effectClient, priority); 6696 lStatus = effect->addHandle(handle); 6697 if (enabled != NULL) { 6698 *enabled = (int)effect->isEnabled(); 6699 } 6700 } 6701 6702Exit: 6703 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6704 Mutex::Autolock _l(mLock); 6705 if (effectCreated) { 6706 chain->removeEffect_l(effect); 6707 } 6708 if (effectRegistered) { 6709 AudioSystem::unregisterEffect(effect->id()); 6710 } 6711 if (chainCreated) { 6712 removeEffectChain_l(chain); 6713 } 6714 handle.clear(); 6715 } 6716 6717 if (status != NULL) { 6718 *status = lStatus; 6719 } 6720 return handle; 6721} 6722 6723sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6724{ 6725 sp<EffectChain> chain = getEffectChain_l(sessionId); 6726 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6727} 6728 6729// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6730// PlaybackThread::mLock held 6731status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6732{ 6733 // check for existing effect chain with the requested audio session 6734 int sessionId = effect->sessionId(); 6735 sp<EffectChain> chain = getEffectChain_l(sessionId); 6736 bool chainCreated = false; 6737 6738 if (chain == 0) { 6739 // create a new chain for this session 6740 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6741 chain = new EffectChain(this, sessionId); 6742 addEffectChain_l(chain); 6743 chain->setStrategy(getStrategyForSession_l(sessionId)); 6744 chainCreated = true; 6745 } 6746 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6747 6748 if (chain->getEffectFromId_l(effect->id()) != 0) { 6749 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6750 this, effect->desc().name, chain.get()); 6751 return BAD_VALUE; 6752 } 6753 6754 status_t status = chain->addEffect_l(effect); 6755 if (status != NO_ERROR) { 6756 if (chainCreated) { 6757 removeEffectChain_l(chain); 6758 } 6759 return status; 6760 } 6761 6762 effect->setDevice(mDevice); 6763 effect->setMode(mAudioFlinger->getMode()); 6764 return NO_ERROR; 6765} 6766 6767void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6768 6769 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6770 effect_descriptor_t desc = effect->desc(); 6771 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6772 detachAuxEffect_l(effect->id()); 6773 } 6774 6775 sp<EffectChain> chain = effect->chain().promote(); 6776 if (chain != 0) { 6777 // remove effect chain if removing last effect 6778 if (chain->removeEffect_l(effect) == 0) { 6779 removeEffectChain_l(chain); 6780 } 6781 } else { 6782 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6783 } 6784} 6785 6786void AudioFlinger::ThreadBase::lockEffectChains_l( 6787 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6788{ 6789 effectChains = mEffectChains; 6790 for (size_t i = 0; i < mEffectChains.size(); i++) { 6791 mEffectChains[i]->lock(); 6792 } 6793} 6794 6795void AudioFlinger::ThreadBase::unlockEffectChains( 6796 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6797{ 6798 for (size_t i = 0; i < effectChains.size(); i++) { 6799 effectChains[i]->unlock(); 6800 } 6801} 6802 6803sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6804{ 6805 Mutex::Autolock _l(mLock); 6806 return getEffectChain_l(sessionId); 6807} 6808 6809sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6810{ 6811 size_t size = mEffectChains.size(); 6812 for (size_t i = 0; i < size; i++) { 6813 if (mEffectChains[i]->sessionId() == sessionId) { 6814 return mEffectChains[i]; 6815 } 6816 } 6817 return 0; 6818} 6819 6820void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6821{ 6822 Mutex::Autolock _l(mLock); 6823 size_t size = mEffectChains.size(); 6824 for (size_t i = 0; i < size; i++) { 6825 mEffectChains[i]->setMode_l(mode); 6826 } 6827} 6828 6829void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6830 const wp<EffectHandle>& handle, 6831 bool unpinIfLast) { 6832 6833 Mutex::Autolock _l(mLock); 6834 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6835 // delete the effect module if removing last handle on it 6836 if (effect->removeHandle(handle) == 0) { 6837 if (!effect->isPinned() || unpinIfLast) { 6838 removeEffect_l(effect); 6839 AudioSystem::unregisterEffect(effect->id()); 6840 } 6841 } 6842} 6843 6844status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6845{ 6846 int session = chain->sessionId(); 6847 int16_t *buffer = mMixBuffer; 6848 bool ownsBuffer = false; 6849 6850 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6851 if (session > 0) { 6852 // Only one effect chain can be present in direct output thread and it uses 6853 // the mix buffer as input 6854 if (mType != DIRECT) { 6855 size_t numSamples = mFrameCount * mChannelCount; 6856 buffer = new int16_t[numSamples]; 6857 memset(buffer, 0, numSamples * sizeof(int16_t)); 6858 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6859 ownsBuffer = true; 6860 } 6861 6862 // Attach all tracks with same session ID to this chain. 6863 for (size_t i = 0; i < mTracks.size(); ++i) { 6864 sp<Track> track = mTracks[i]; 6865 if (session == track->sessionId()) { 6866 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6867 track->setMainBuffer(buffer); 6868 chain->incTrackCnt(); 6869 } 6870 } 6871 6872 // indicate all active tracks in the chain 6873 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6874 sp<Track> track = mActiveTracks[i].promote(); 6875 if (track == 0) continue; 6876 if (session == track->sessionId()) { 6877 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6878 chain->incActiveTrackCnt(); 6879 } 6880 } 6881 } 6882 6883 chain->setInBuffer(buffer, ownsBuffer); 6884 chain->setOutBuffer(mMixBuffer); 6885 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6886 // chains list in order to be processed last as it contains output stage effects 6887 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6888 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6889 // after track specific effects and before output stage 6890 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6891 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6892 // Effect chain for other sessions are inserted at beginning of effect 6893 // chains list to be processed before output mix effects. Relative order between other 6894 // sessions is not important 6895 size_t size = mEffectChains.size(); 6896 size_t i = 0; 6897 for (i = 0; i < size; i++) { 6898 if (mEffectChains[i]->sessionId() < session) break; 6899 } 6900 mEffectChains.insertAt(chain, i); 6901 checkSuspendOnAddEffectChain_l(chain); 6902 6903 return NO_ERROR; 6904} 6905 6906size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6907{ 6908 int session = chain->sessionId(); 6909 6910 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6911 6912 for (size_t i = 0; i < mEffectChains.size(); i++) { 6913 if (chain == mEffectChains[i]) { 6914 mEffectChains.removeAt(i); 6915 // detach all active tracks from the chain 6916 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6917 sp<Track> track = mActiveTracks[i].promote(); 6918 if (track == 0) continue; 6919 if (session == track->sessionId()) { 6920 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6921 chain.get(), session); 6922 chain->decActiveTrackCnt(); 6923 } 6924 } 6925 6926 // detach all tracks with same session ID from this chain 6927 for (size_t i = 0; i < mTracks.size(); ++i) { 6928 sp<Track> track = mTracks[i]; 6929 if (session == track->sessionId()) { 6930 track->setMainBuffer(mMixBuffer); 6931 chain->decTrackCnt(); 6932 } 6933 } 6934 break; 6935 } 6936 } 6937 return mEffectChains.size(); 6938} 6939 6940status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6941 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6942{ 6943 Mutex::Autolock _l(mLock); 6944 return attachAuxEffect_l(track, EffectId); 6945} 6946 6947status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6948 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6949{ 6950 status_t status = NO_ERROR; 6951 6952 if (EffectId == 0) { 6953 track->setAuxBuffer(0, NULL); 6954 } else { 6955 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6956 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6957 if (effect != 0) { 6958 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6959 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6960 } else { 6961 status = INVALID_OPERATION; 6962 } 6963 } else { 6964 status = BAD_VALUE; 6965 } 6966 } 6967 return status; 6968} 6969 6970void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6971{ 6972 for (size_t i = 0; i < mTracks.size(); ++i) { 6973 sp<Track> track = mTracks[i]; 6974 if (track->auxEffectId() == effectId) { 6975 attachAuxEffect_l(track, 0); 6976 } 6977 } 6978} 6979 6980status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6981{ 6982 // only one chain per input thread 6983 if (mEffectChains.size() != 0) { 6984 return INVALID_OPERATION; 6985 } 6986 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6987 6988 chain->setInBuffer(NULL); 6989 chain->setOutBuffer(NULL); 6990 6991 checkSuspendOnAddEffectChain_l(chain); 6992 6993 mEffectChains.add(chain); 6994 6995 return NO_ERROR; 6996} 6997 6998size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6999{ 7000 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7001 ALOGW_IF(mEffectChains.size() != 1, 7002 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7003 chain.get(), mEffectChains.size(), this); 7004 if (mEffectChains.size() == 1) { 7005 mEffectChains.removeAt(0); 7006 } 7007 return 0; 7008} 7009 7010// ---------------------------------------------------------------------------- 7011// EffectModule implementation 7012// ---------------------------------------------------------------------------- 7013 7014#undef LOG_TAG 7015#define LOG_TAG "AudioFlinger::EffectModule" 7016 7017AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7018 const wp<AudioFlinger::EffectChain>& chain, 7019 effect_descriptor_t *desc, 7020 int id, 7021 int sessionId) 7022 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7023 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7024{ 7025 ALOGV("Constructor %p", this); 7026 int lStatus; 7027 if (thread == NULL) { 7028 return; 7029 } 7030 7031 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7032 7033 // create effect engine from effect factory 7034 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7035 7036 if (mStatus != NO_ERROR) { 7037 return; 7038 } 7039 lStatus = init(); 7040 if (lStatus < 0) { 7041 mStatus = lStatus; 7042 goto Error; 7043 } 7044 7045 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7046 mPinned = true; 7047 } 7048 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7049 return; 7050Error: 7051 EffectRelease(mEffectInterface); 7052 mEffectInterface = NULL; 7053 ALOGV("Constructor Error %d", mStatus); 7054} 7055 7056AudioFlinger::EffectModule::~EffectModule() 7057{ 7058 ALOGV("Destructor %p", this); 7059 if (mEffectInterface != NULL) { 7060 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7061 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7062 sp<ThreadBase> thread = mThread.promote(); 7063 if (thread != 0) { 7064 audio_stream_t *stream = thread->stream(); 7065 if (stream != NULL) { 7066 stream->remove_audio_effect(stream, mEffectInterface); 7067 } 7068 } 7069 } 7070 // release effect engine 7071 EffectRelease(mEffectInterface); 7072 } 7073} 7074 7075status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7076{ 7077 status_t status; 7078 7079 Mutex::Autolock _l(mLock); 7080 int priority = handle->priority(); 7081 size_t size = mHandles.size(); 7082 sp<EffectHandle> h; 7083 size_t i; 7084 for (i = 0; i < size; i++) { 7085 h = mHandles[i].promote(); 7086 if (h == 0) continue; 7087 if (h->priority() <= priority) break; 7088 } 7089 // if inserted in first place, move effect control from previous owner to this handle 7090 if (i == 0) { 7091 bool enabled = false; 7092 if (h != 0) { 7093 enabled = h->enabled(); 7094 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7095 } 7096 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7097 status = NO_ERROR; 7098 } else { 7099 status = ALREADY_EXISTS; 7100 } 7101 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7102 mHandles.insertAt(handle, i); 7103 return status; 7104} 7105 7106size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7107{ 7108 Mutex::Autolock _l(mLock); 7109 size_t size = mHandles.size(); 7110 size_t i; 7111 for (i = 0; i < size; i++) { 7112 if (mHandles[i] == handle) break; 7113 } 7114 if (i == size) { 7115 return size; 7116 } 7117 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7118 7119 bool enabled = false; 7120 EffectHandle *hdl = handle.unsafe_get(); 7121 if (hdl != NULL) { 7122 ALOGV("removeHandle() unsafe_get OK"); 7123 enabled = hdl->enabled(); 7124 } 7125 mHandles.removeAt(i); 7126 size = mHandles.size(); 7127 // if removed from first place, move effect control from this handle to next in line 7128 if (i == 0 && size != 0) { 7129 sp<EffectHandle> h = mHandles[0].promote(); 7130 if (h != 0) { 7131 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7132 } 7133 } 7134 7135 // Prevent calls to process() and other functions on effect interface from now on. 7136 // The effect engine will be released by the destructor when the last strong reference on 7137 // this object is released which can happen after next process is called. 7138 if (size == 0 && !mPinned) { 7139 mState = DESTROYED; 7140 } 7141 7142 return size; 7143} 7144 7145sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7146{ 7147 Mutex::Autolock _l(mLock); 7148 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7149} 7150 7151void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7152{ 7153 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7154 // keep a strong reference on this EffectModule to avoid calling the 7155 // destructor before we exit 7156 sp<EffectModule> keep(this); 7157 { 7158 sp<ThreadBase> thread = mThread.promote(); 7159 if (thread != 0) { 7160 thread->disconnectEffect(keep, handle, unpinIfLast); 7161 } 7162 } 7163} 7164 7165void AudioFlinger::EffectModule::updateState() { 7166 Mutex::Autolock _l(mLock); 7167 7168 switch (mState) { 7169 case RESTART: 7170 reset_l(); 7171 // FALL THROUGH 7172 7173 case STARTING: 7174 // clear auxiliary effect input buffer for next accumulation 7175 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7176 memset(mConfig.inputCfg.buffer.raw, 7177 0, 7178 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7179 } 7180 start_l(); 7181 mState = ACTIVE; 7182 break; 7183 case STOPPING: 7184 stop_l(); 7185 mDisableWaitCnt = mMaxDisableWaitCnt; 7186 mState = STOPPED; 7187 break; 7188 case STOPPED: 7189 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7190 // turn off sequence. 7191 if (--mDisableWaitCnt == 0) { 7192 reset_l(); 7193 mState = IDLE; 7194 } 7195 break; 7196 default: //IDLE , ACTIVE, DESTROYED 7197 break; 7198 } 7199} 7200 7201void AudioFlinger::EffectModule::process() 7202{ 7203 Mutex::Autolock _l(mLock); 7204 7205 if (mState == DESTROYED || mEffectInterface == NULL || 7206 mConfig.inputCfg.buffer.raw == NULL || 7207 mConfig.outputCfg.buffer.raw == NULL) { 7208 return; 7209 } 7210 7211 if (isProcessEnabled()) { 7212 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7213 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7214 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7215 mConfig.inputCfg.buffer.s32, 7216 mConfig.inputCfg.buffer.frameCount/2); 7217 } 7218 7219 // do the actual processing in the effect engine 7220 int ret = (*mEffectInterface)->process(mEffectInterface, 7221 &mConfig.inputCfg.buffer, 7222 &mConfig.outputCfg.buffer); 7223 7224 // force transition to IDLE state when engine is ready 7225 if (mState == STOPPED && ret == -ENODATA) { 7226 mDisableWaitCnt = 1; 7227 } 7228 7229 // clear auxiliary effect input buffer for next accumulation 7230 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7231 memset(mConfig.inputCfg.buffer.raw, 0, 7232 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7233 } 7234 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7235 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7236 // If an insert effect is idle and input buffer is different from output buffer, 7237 // accumulate input onto output 7238 sp<EffectChain> chain = mChain.promote(); 7239 if (chain != 0 && chain->activeTrackCnt() != 0) { 7240 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7241 int16_t *in = mConfig.inputCfg.buffer.s16; 7242 int16_t *out = mConfig.outputCfg.buffer.s16; 7243 for (size_t i = 0; i < frameCnt; i++) { 7244 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7245 } 7246 } 7247 } 7248} 7249 7250void AudioFlinger::EffectModule::reset_l() 7251{ 7252 if (mEffectInterface == NULL) { 7253 return; 7254 } 7255 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7256} 7257 7258status_t AudioFlinger::EffectModule::configure() 7259{ 7260 uint32_t channels; 7261 if (mEffectInterface == NULL) { 7262 return NO_INIT; 7263 } 7264 7265 sp<ThreadBase> thread = mThread.promote(); 7266 if (thread == 0) { 7267 return DEAD_OBJECT; 7268 } 7269 7270 // TODO: handle configuration of effects replacing track process 7271 if (thread->channelCount() == 1) { 7272 channels = AUDIO_CHANNEL_OUT_MONO; 7273 } else { 7274 channels = AUDIO_CHANNEL_OUT_STEREO; 7275 } 7276 7277 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7278 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7279 } else { 7280 mConfig.inputCfg.channels = channels; 7281 } 7282 mConfig.outputCfg.channels = channels; 7283 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7284 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7285 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7286 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7287 mConfig.inputCfg.bufferProvider.cookie = NULL; 7288 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7289 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7290 mConfig.outputCfg.bufferProvider.cookie = NULL; 7291 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7292 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7293 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7294 // Insert effect: 7295 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7296 // always overwrites output buffer: input buffer == output buffer 7297 // - in other sessions: 7298 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7299 // other effect: overwrites output buffer: input buffer == output buffer 7300 // Auxiliary effect: 7301 // accumulates in output buffer: input buffer != output buffer 7302 // Therefore: accumulate <=> input buffer != output buffer 7303 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7304 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7305 } else { 7306 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7307 } 7308 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7309 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7310 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7311 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7312 7313 ALOGV("configure() %p thread %p buffer %p framecount %d", 7314 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7315 7316 status_t cmdStatus; 7317 uint32_t size = sizeof(int); 7318 status_t status = (*mEffectInterface)->command(mEffectInterface, 7319 EFFECT_CMD_SET_CONFIG, 7320 sizeof(effect_config_t), 7321 &mConfig, 7322 &size, 7323 &cmdStatus); 7324 if (status == 0) { 7325 status = cmdStatus; 7326 } 7327 7328 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7329 (1000 * mConfig.outputCfg.buffer.frameCount); 7330 7331 return status; 7332} 7333 7334status_t AudioFlinger::EffectModule::init() 7335{ 7336 Mutex::Autolock _l(mLock); 7337 if (mEffectInterface == NULL) { 7338 return NO_INIT; 7339 } 7340 status_t cmdStatus; 7341 uint32_t size = sizeof(status_t); 7342 status_t status = (*mEffectInterface)->command(mEffectInterface, 7343 EFFECT_CMD_INIT, 7344 0, 7345 NULL, 7346 &size, 7347 &cmdStatus); 7348 if (status == 0) { 7349 status = cmdStatus; 7350 } 7351 return status; 7352} 7353 7354status_t AudioFlinger::EffectModule::start() 7355{ 7356 Mutex::Autolock _l(mLock); 7357 return start_l(); 7358} 7359 7360status_t AudioFlinger::EffectModule::start_l() 7361{ 7362 if (mEffectInterface == NULL) { 7363 return NO_INIT; 7364 } 7365 status_t cmdStatus; 7366 uint32_t size = sizeof(status_t); 7367 status_t status = (*mEffectInterface)->command(mEffectInterface, 7368 EFFECT_CMD_ENABLE, 7369 0, 7370 NULL, 7371 &size, 7372 &cmdStatus); 7373 if (status == 0) { 7374 status = cmdStatus; 7375 } 7376 if (status == 0 && 7377 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7378 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7379 sp<ThreadBase> thread = mThread.promote(); 7380 if (thread != 0) { 7381 audio_stream_t *stream = thread->stream(); 7382 if (stream != NULL) { 7383 stream->add_audio_effect(stream, mEffectInterface); 7384 } 7385 } 7386 } 7387 return status; 7388} 7389 7390status_t AudioFlinger::EffectModule::stop() 7391{ 7392 Mutex::Autolock _l(mLock); 7393 return stop_l(); 7394} 7395 7396status_t AudioFlinger::EffectModule::stop_l() 7397{ 7398 if (mEffectInterface == NULL) { 7399 return NO_INIT; 7400 } 7401 status_t cmdStatus; 7402 uint32_t size = sizeof(status_t); 7403 status_t status = (*mEffectInterface)->command(mEffectInterface, 7404 EFFECT_CMD_DISABLE, 7405 0, 7406 NULL, 7407 &size, 7408 &cmdStatus); 7409 if (status == 0) { 7410 status = cmdStatus; 7411 } 7412 if (status == 0 && 7413 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7414 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7415 sp<ThreadBase> thread = mThread.promote(); 7416 if (thread != 0) { 7417 audio_stream_t *stream = thread->stream(); 7418 if (stream != NULL) { 7419 stream->remove_audio_effect(stream, mEffectInterface); 7420 } 7421 } 7422 } 7423 return status; 7424} 7425 7426status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7427 uint32_t cmdSize, 7428 void *pCmdData, 7429 uint32_t *replySize, 7430 void *pReplyData) 7431{ 7432 Mutex::Autolock _l(mLock); 7433// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7434 7435 if (mState == DESTROYED || mEffectInterface == NULL) { 7436 return NO_INIT; 7437 } 7438 status_t status = (*mEffectInterface)->command(mEffectInterface, 7439 cmdCode, 7440 cmdSize, 7441 pCmdData, 7442 replySize, 7443 pReplyData); 7444 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7445 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7446 for (size_t i = 1; i < mHandles.size(); i++) { 7447 sp<EffectHandle> h = mHandles[i].promote(); 7448 if (h != 0) { 7449 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7450 } 7451 } 7452 } 7453 return status; 7454} 7455 7456status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7457{ 7458 7459 Mutex::Autolock _l(mLock); 7460 ALOGV("setEnabled %p enabled %d", this, enabled); 7461 7462 if (enabled != isEnabled()) { 7463 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7464 if (enabled && status != NO_ERROR) { 7465 return status; 7466 } 7467 7468 switch (mState) { 7469 // going from disabled to enabled 7470 case IDLE: 7471 mState = STARTING; 7472 break; 7473 case STOPPED: 7474 mState = RESTART; 7475 break; 7476 case STOPPING: 7477 mState = ACTIVE; 7478 break; 7479 7480 // going from enabled to disabled 7481 case RESTART: 7482 mState = STOPPED; 7483 break; 7484 case STARTING: 7485 mState = IDLE; 7486 break; 7487 case ACTIVE: 7488 mState = STOPPING; 7489 break; 7490 case DESTROYED: 7491 return NO_ERROR; // simply ignore as we are being destroyed 7492 } 7493 for (size_t i = 1; i < mHandles.size(); i++) { 7494 sp<EffectHandle> h = mHandles[i].promote(); 7495 if (h != 0) { 7496 h->setEnabled(enabled); 7497 } 7498 } 7499 } 7500 return NO_ERROR; 7501} 7502 7503bool AudioFlinger::EffectModule::isEnabled() const 7504{ 7505 switch (mState) { 7506 case RESTART: 7507 case STARTING: 7508 case ACTIVE: 7509 return true; 7510 case IDLE: 7511 case STOPPING: 7512 case STOPPED: 7513 case DESTROYED: 7514 default: 7515 return false; 7516 } 7517} 7518 7519bool AudioFlinger::EffectModule::isProcessEnabled() const 7520{ 7521 switch (mState) { 7522 case RESTART: 7523 case ACTIVE: 7524 case STOPPING: 7525 case STOPPED: 7526 return true; 7527 case IDLE: 7528 case STARTING: 7529 case DESTROYED: 7530 default: 7531 return false; 7532 } 7533} 7534 7535status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7536{ 7537 Mutex::Autolock _l(mLock); 7538 status_t status = NO_ERROR; 7539 7540 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7541 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7542 if (isProcessEnabled() && 7543 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7544 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7545 status_t cmdStatus; 7546 uint32_t volume[2]; 7547 uint32_t *pVolume = NULL; 7548 uint32_t size = sizeof(volume); 7549 volume[0] = *left; 7550 volume[1] = *right; 7551 if (controller) { 7552 pVolume = volume; 7553 } 7554 status = (*mEffectInterface)->command(mEffectInterface, 7555 EFFECT_CMD_SET_VOLUME, 7556 size, 7557 volume, 7558 &size, 7559 pVolume); 7560 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7561 *left = volume[0]; 7562 *right = volume[1]; 7563 } 7564 } 7565 return status; 7566} 7567 7568status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7569{ 7570 Mutex::Autolock _l(mLock); 7571 status_t status = NO_ERROR; 7572 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7573 // audio pre processing modules on RecordThread can receive both output and 7574 // input device indication in the same call 7575 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7576 if (dev) { 7577 status_t cmdStatus; 7578 uint32_t size = sizeof(status_t); 7579 7580 status = (*mEffectInterface)->command(mEffectInterface, 7581 EFFECT_CMD_SET_DEVICE, 7582 sizeof(uint32_t), 7583 &dev, 7584 &size, 7585 &cmdStatus); 7586 if (status == NO_ERROR) { 7587 status = cmdStatus; 7588 } 7589 } 7590 dev = device & AUDIO_DEVICE_IN_ALL; 7591 if (dev) { 7592 status_t cmdStatus; 7593 uint32_t size = sizeof(status_t); 7594 7595 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7596 EFFECT_CMD_SET_INPUT_DEVICE, 7597 sizeof(uint32_t), 7598 &dev, 7599 &size, 7600 &cmdStatus); 7601 if (status2 == NO_ERROR) { 7602 status2 = cmdStatus; 7603 } 7604 if (status == NO_ERROR) { 7605 status = status2; 7606 } 7607 } 7608 } 7609 return status; 7610} 7611 7612status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7613{ 7614 Mutex::Autolock _l(mLock); 7615 status_t status = NO_ERROR; 7616 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7617 status_t cmdStatus; 7618 uint32_t size = sizeof(status_t); 7619 status = (*mEffectInterface)->command(mEffectInterface, 7620 EFFECT_CMD_SET_AUDIO_MODE, 7621 sizeof(audio_mode_t), 7622 &mode, 7623 &size, 7624 &cmdStatus); 7625 if (status == NO_ERROR) { 7626 status = cmdStatus; 7627 } 7628 } 7629 return status; 7630} 7631 7632void AudioFlinger::EffectModule::setSuspended(bool suspended) 7633{ 7634 Mutex::Autolock _l(mLock); 7635 mSuspended = suspended; 7636} 7637 7638bool AudioFlinger::EffectModule::suspended() const 7639{ 7640 Mutex::Autolock _l(mLock); 7641 return mSuspended; 7642} 7643 7644status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7645{ 7646 const size_t SIZE = 256; 7647 char buffer[SIZE]; 7648 String8 result; 7649 7650 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7651 result.append(buffer); 7652 7653 bool locked = tryLock(mLock); 7654 // failed to lock - AudioFlinger is probably deadlocked 7655 if (!locked) { 7656 result.append("\t\tCould not lock Fx mutex:\n"); 7657 } 7658 7659 result.append("\t\tSession Status State Engine:\n"); 7660 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7661 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7662 result.append(buffer); 7663 7664 result.append("\t\tDescriptor:\n"); 7665 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7666 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7667 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7668 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7669 result.append(buffer); 7670 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7671 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7672 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7673 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7674 result.append(buffer); 7675 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7676 mDescriptor.apiVersion, 7677 mDescriptor.flags); 7678 result.append(buffer); 7679 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7680 mDescriptor.name); 7681 result.append(buffer); 7682 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7683 mDescriptor.implementor); 7684 result.append(buffer); 7685 7686 result.append("\t\t- Input configuration:\n"); 7687 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7688 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7689 (uint32_t)mConfig.inputCfg.buffer.raw, 7690 mConfig.inputCfg.buffer.frameCount, 7691 mConfig.inputCfg.samplingRate, 7692 mConfig.inputCfg.channels, 7693 mConfig.inputCfg.format); 7694 result.append(buffer); 7695 7696 result.append("\t\t- Output configuration:\n"); 7697 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7698 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7699 (uint32_t)mConfig.outputCfg.buffer.raw, 7700 mConfig.outputCfg.buffer.frameCount, 7701 mConfig.outputCfg.samplingRate, 7702 mConfig.outputCfg.channels, 7703 mConfig.outputCfg.format); 7704 result.append(buffer); 7705 7706 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7707 result.append(buffer); 7708 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7709 for (size_t i = 0; i < mHandles.size(); ++i) { 7710 sp<EffectHandle> handle = mHandles[i].promote(); 7711 if (handle != 0) { 7712 handle->dump(buffer, SIZE); 7713 result.append(buffer); 7714 } 7715 } 7716 7717 result.append("\n"); 7718 7719 write(fd, result.string(), result.length()); 7720 7721 if (locked) { 7722 mLock.unlock(); 7723 } 7724 7725 return NO_ERROR; 7726} 7727 7728// ---------------------------------------------------------------------------- 7729// EffectHandle implementation 7730// ---------------------------------------------------------------------------- 7731 7732#undef LOG_TAG 7733#define LOG_TAG "AudioFlinger::EffectHandle" 7734 7735AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7736 const sp<AudioFlinger::Client>& client, 7737 const sp<IEffectClient>& effectClient, 7738 int32_t priority) 7739 : BnEffect(), 7740 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7741 mPriority(priority), mHasControl(false), mEnabled(false) 7742{ 7743 ALOGV("constructor %p", this); 7744 7745 if (client == 0) { 7746 return; 7747 } 7748 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7749 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7750 if (mCblkMemory != 0) { 7751 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7752 7753 if (mCblk != NULL) { 7754 new(mCblk) effect_param_cblk_t(); 7755 mBuffer = (uint8_t *)mCblk + bufOffset; 7756 } 7757 } else { 7758 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7759 return; 7760 } 7761} 7762 7763AudioFlinger::EffectHandle::~EffectHandle() 7764{ 7765 ALOGV("Destructor %p", this); 7766 disconnect(false); 7767 ALOGV("Destructor DONE %p", this); 7768} 7769 7770status_t AudioFlinger::EffectHandle::enable() 7771{ 7772 ALOGV("enable %p", this); 7773 if (!mHasControl) return INVALID_OPERATION; 7774 if (mEffect == 0) return DEAD_OBJECT; 7775 7776 if (mEnabled) { 7777 return NO_ERROR; 7778 } 7779 7780 mEnabled = true; 7781 7782 sp<ThreadBase> thread = mEffect->thread().promote(); 7783 if (thread != 0) { 7784 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7785 } 7786 7787 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7788 if (mEffect->suspended()) { 7789 return NO_ERROR; 7790 } 7791 7792 status_t status = mEffect->setEnabled(true); 7793 if (status != NO_ERROR) { 7794 if (thread != 0) { 7795 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7796 } 7797 mEnabled = false; 7798 } 7799 return status; 7800} 7801 7802status_t AudioFlinger::EffectHandle::disable() 7803{ 7804 ALOGV("disable %p", this); 7805 if (!mHasControl) return INVALID_OPERATION; 7806 if (mEffect == 0) return DEAD_OBJECT; 7807 7808 if (!mEnabled) { 7809 return NO_ERROR; 7810 } 7811 mEnabled = false; 7812 7813 if (mEffect->suspended()) { 7814 return NO_ERROR; 7815 } 7816 7817 status_t status = mEffect->setEnabled(false); 7818 7819 sp<ThreadBase> thread = mEffect->thread().promote(); 7820 if (thread != 0) { 7821 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7822 } 7823 7824 return status; 7825} 7826 7827void AudioFlinger::EffectHandle::disconnect() 7828{ 7829 disconnect(true); 7830} 7831 7832void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7833{ 7834 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7835 if (mEffect == 0) { 7836 return; 7837 } 7838 mEffect->disconnect(this, unpinIfLast); 7839 7840 if (mHasControl && mEnabled) { 7841 sp<ThreadBase> thread = mEffect->thread().promote(); 7842 if (thread != 0) { 7843 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7844 } 7845 } 7846 7847 // release sp on module => module destructor can be called now 7848 mEffect.clear(); 7849 if (mClient != 0) { 7850 if (mCblk != NULL) { 7851 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7852 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7853 } 7854 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7855 // Client destructor must run with AudioFlinger mutex locked 7856 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7857 mClient.clear(); 7858 } 7859} 7860 7861status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7862 uint32_t cmdSize, 7863 void *pCmdData, 7864 uint32_t *replySize, 7865 void *pReplyData) 7866{ 7867// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7868// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7869 7870 // only get parameter command is permitted for applications not controlling the effect 7871 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7872 return INVALID_OPERATION; 7873 } 7874 if (mEffect == 0) return DEAD_OBJECT; 7875 if (mClient == 0) return INVALID_OPERATION; 7876 7877 // handle commands that are not forwarded transparently to effect engine 7878 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7879 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7880 // no risk to block the whole media server process or mixer threads is we are stuck here 7881 Mutex::Autolock _l(mCblk->lock); 7882 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7883 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7884 mCblk->serverIndex = 0; 7885 mCblk->clientIndex = 0; 7886 return BAD_VALUE; 7887 } 7888 status_t status = NO_ERROR; 7889 while (mCblk->serverIndex < mCblk->clientIndex) { 7890 int reply; 7891 uint32_t rsize = sizeof(int); 7892 int *p = (int *)(mBuffer + mCblk->serverIndex); 7893 int size = *p++; 7894 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7895 ALOGW("command(): invalid parameter block size"); 7896 break; 7897 } 7898 effect_param_t *param = (effect_param_t *)p; 7899 if (param->psize == 0 || param->vsize == 0) { 7900 ALOGW("command(): null parameter or value size"); 7901 mCblk->serverIndex += size; 7902 continue; 7903 } 7904 uint32_t psize = sizeof(effect_param_t) + 7905 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7906 param->vsize; 7907 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7908 psize, 7909 p, 7910 &rsize, 7911 &reply); 7912 // stop at first error encountered 7913 if (ret != NO_ERROR) { 7914 status = ret; 7915 *(int *)pReplyData = reply; 7916 break; 7917 } else if (reply != NO_ERROR) { 7918 *(int *)pReplyData = reply; 7919 break; 7920 } 7921 mCblk->serverIndex += size; 7922 } 7923 mCblk->serverIndex = 0; 7924 mCblk->clientIndex = 0; 7925 return status; 7926 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7927 *(int *)pReplyData = NO_ERROR; 7928 return enable(); 7929 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7930 *(int *)pReplyData = NO_ERROR; 7931 return disable(); 7932 } 7933 7934 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7935} 7936 7937void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7938{ 7939 ALOGV("setControl %p control %d", this, hasControl); 7940 7941 mHasControl = hasControl; 7942 mEnabled = enabled; 7943 7944 if (signal && mEffectClient != 0) { 7945 mEffectClient->controlStatusChanged(hasControl); 7946 } 7947} 7948 7949void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7950 uint32_t cmdSize, 7951 void *pCmdData, 7952 uint32_t replySize, 7953 void *pReplyData) 7954{ 7955 if (mEffectClient != 0) { 7956 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7957 } 7958} 7959 7960 7961 7962void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7963{ 7964 if (mEffectClient != 0) { 7965 mEffectClient->enableStatusChanged(enabled); 7966 } 7967} 7968 7969status_t AudioFlinger::EffectHandle::onTransact( 7970 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7971{ 7972 return BnEffect::onTransact(code, data, reply, flags); 7973} 7974 7975 7976void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7977{ 7978 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7979 7980 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7981 (mClient == 0) ? getpid_cached : mClient->pid(), 7982 mPriority, 7983 mHasControl, 7984 !locked, 7985 mCblk ? mCblk->clientIndex : 0, 7986 mCblk ? mCblk->serverIndex : 0 7987 ); 7988 7989 if (locked) { 7990 mCblk->lock.unlock(); 7991 } 7992} 7993 7994#undef LOG_TAG 7995#define LOG_TAG "AudioFlinger::EffectChain" 7996 7997AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7998 int sessionId) 7999 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8000 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8001 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8002{ 8003 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8004 if (thread == NULL) { 8005 return; 8006 } 8007 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8008 thread->frameCount(); 8009} 8010 8011AudioFlinger::EffectChain::~EffectChain() 8012{ 8013 if (mOwnInBuffer) { 8014 delete mInBuffer; 8015 } 8016 8017} 8018 8019// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8020sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8021{ 8022 size_t size = mEffects.size(); 8023 8024 for (size_t i = 0; i < size; i++) { 8025 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8026 return mEffects[i]; 8027 } 8028 } 8029 return 0; 8030} 8031 8032// getEffectFromId_l() must be called with ThreadBase::mLock held 8033sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8034{ 8035 size_t size = mEffects.size(); 8036 8037 for (size_t i = 0; i < size; i++) { 8038 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8039 if (id == 0 || mEffects[i]->id() == id) { 8040 return mEffects[i]; 8041 } 8042 } 8043 return 0; 8044} 8045 8046// getEffectFromType_l() must be called with ThreadBase::mLock held 8047sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8048 const effect_uuid_t *type) 8049{ 8050 size_t size = mEffects.size(); 8051 8052 for (size_t i = 0; i < size; i++) { 8053 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8054 return mEffects[i]; 8055 } 8056 } 8057 return 0; 8058} 8059 8060// Must be called with EffectChain::mLock locked 8061void AudioFlinger::EffectChain::process_l() 8062{ 8063 sp<ThreadBase> thread = mThread.promote(); 8064 if (thread == 0) { 8065 ALOGW("process_l(): cannot promote mixer thread"); 8066 return; 8067 } 8068 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8069 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8070 // always process effects unless no more tracks are on the session and the effect tail 8071 // has been rendered 8072 bool doProcess = true; 8073 if (!isGlobalSession) { 8074 bool tracksOnSession = (trackCnt() != 0); 8075 8076 if (!tracksOnSession && mTailBufferCount == 0) { 8077 doProcess = false; 8078 } 8079 8080 if (activeTrackCnt() == 0) { 8081 // if no track is active and the effect tail has not been rendered, 8082 // the input buffer must be cleared here as the mixer process will not do it 8083 if (tracksOnSession || mTailBufferCount > 0) { 8084 size_t numSamples = thread->frameCount() * thread->channelCount(); 8085 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8086 if (mTailBufferCount > 0) { 8087 mTailBufferCount--; 8088 } 8089 } 8090 } 8091 } 8092 8093 size_t size = mEffects.size(); 8094 if (doProcess) { 8095 for (size_t i = 0; i < size; i++) { 8096 mEffects[i]->process(); 8097 } 8098 } 8099 for (size_t i = 0; i < size; i++) { 8100 mEffects[i]->updateState(); 8101 } 8102} 8103 8104// addEffect_l() must be called with PlaybackThread::mLock held 8105status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8106{ 8107 effect_descriptor_t desc = effect->desc(); 8108 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8109 8110 Mutex::Autolock _l(mLock); 8111 effect->setChain(this); 8112 sp<ThreadBase> thread = mThread.promote(); 8113 if (thread == 0) { 8114 return NO_INIT; 8115 } 8116 effect->setThread(thread); 8117 8118 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8119 // Auxiliary effects are inserted at the beginning of mEffects vector as 8120 // they are processed first and accumulated in chain input buffer 8121 mEffects.insertAt(effect, 0); 8122 8123 // the input buffer for auxiliary effect contains mono samples in 8124 // 32 bit format. This is to avoid saturation in AudoMixer 8125 // accumulation stage. Saturation is done in EffectModule::process() before 8126 // calling the process in effect engine 8127 size_t numSamples = thread->frameCount(); 8128 int32_t *buffer = new int32_t[numSamples]; 8129 memset(buffer, 0, numSamples * sizeof(int32_t)); 8130 effect->setInBuffer((int16_t *)buffer); 8131 // auxiliary effects output samples to chain input buffer for further processing 8132 // by insert effects 8133 effect->setOutBuffer(mInBuffer); 8134 } else { 8135 // Insert effects are inserted at the end of mEffects vector as they are processed 8136 // after track and auxiliary effects. 8137 // Insert effect order as a function of indicated preference: 8138 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8139 // another effect is present 8140 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8141 // last effect claiming first position 8142 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8143 // first effect claiming last position 8144 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8145 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8146 // already present 8147 8148 size_t size = mEffects.size(); 8149 size_t idx_insert = size; 8150 ssize_t idx_insert_first = -1; 8151 ssize_t idx_insert_last = -1; 8152 8153 for (size_t i = 0; i < size; i++) { 8154 effect_descriptor_t d = mEffects[i]->desc(); 8155 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8156 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8157 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8158 // check invalid effect chaining combinations 8159 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8160 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8161 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8162 return INVALID_OPERATION; 8163 } 8164 // remember position of first insert effect and by default 8165 // select this as insert position for new effect 8166 if (idx_insert == size) { 8167 idx_insert = i; 8168 } 8169 // remember position of last insert effect claiming 8170 // first position 8171 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8172 idx_insert_first = i; 8173 } 8174 // remember position of first insert effect claiming 8175 // last position 8176 if (iPref == EFFECT_FLAG_INSERT_LAST && 8177 idx_insert_last == -1) { 8178 idx_insert_last = i; 8179 } 8180 } 8181 } 8182 8183 // modify idx_insert from first position if needed 8184 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8185 if (idx_insert_last != -1) { 8186 idx_insert = idx_insert_last; 8187 } else { 8188 idx_insert = size; 8189 } 8190 } else { 8191 if (idx_insert_first != -1) { 8192 idx_insert = idx_insert_first + 1; 8193 } 8194 } 8195 8196 // always read samples from chain input buffer 8197 effect->setInBuffer(mInBuffer); 8198 8199 // if last effect in the chain, output samples to chain 8200 // output buffer, otherwise to chain input buffer 8201 if (idx_insert == size) { 8202 if (idx_insert != 0) { 8203 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8204 mEffects[idx_insert-1]->configure(); 8205 } 8206 effect->setOutBuffer(mOutBuffer); 8207 } else { 8208 effect->setOutBuffer(mInBuffer); 8209 } 8210 mEffects.insertAt(effect, idx_insert); 8211 8212 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8213 } 8214 effect->configure(); 8215 return NO_ERROR; 8216} 8217 8218// removeEffect_l() must be called with PlaybackThread::mLock held 8219size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8220{ 8221 Mutex::Autolock _l(mLock); 8222 size_t size = mEffects.size(); 8223 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8224 8225 for (size_t i = 0; i < size; i++) { 8226 if (effect == mEffects[i]) { 8227 // calling stop here will remove pre-processing effect from the audio HAL. 8228 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8229 // the middle of a read from audio HAL 8230 if (mEffects[i]->state() == EffectModule::ACTIVE || 8231 mEffects[i]->state() == EffectModule::STOPPING) { 8232 mEffects[i]->stop(); 8233 } 8234 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8235 delete[] effect->inBuffer(); 8236 } else { 8237 if (i == size - 1 && i != 0) { 8238 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8239 mEffects[i - 1]->configure(); 8240 } 8241 } 8242 mEffects.removeAt(i); 8243 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8244 break; 8245 } 8246 } 8247 8248 return mEffects.size(); 8249} 8250 8251// setDevice_l() must be called with PlaybackThread::mLock held 8252void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8253{ 8254 size_t size = mEffects.size(); 8255 for (size_t i = 0; i < size; i++) { 8256 mEffects[i]->setDevice(device); 8257 } 8258} 8259 8260// setMode_l() must be called with PlaybackThread::mLock held 8261void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8262{ 8263 size_t size = mEffects.size(); 8264 for (size_t i = 0; i < size; i++) { 8265 mEffects[i]->setMode(mode); 8266 } 8267} 8268 8269// setVolume_l() must be called with PlaybackThread::mLock held 8270bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8271{ 8272 uint32_t newLeft = *left; 8273 uint32_t newRight = *right; 8274 bool hasControl = false; 8275 int ctrlIdx = -1; 8276 size_t size = mEffects.size(); 8277 8278 // first update volume controller 8279 for (size_t i = size; i > 0; i--) { 8280 if (mEffects[i - 1]->isProcessEnabled() && 8281 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8282 ctrlIdx = i - 1; 8283 hasControl = true; 8284 break; 8285 } 8286 } 8287 8288 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8289 if (hasControl) { 8290 *left = mNewLeftVolume; 8291 *right = mNewRightVolume; 8292 } 8293 return hasControl; 8294 } 8295 8296 mVolumeCtrlIdx = ctrlIdx; 8297 mLeftVolume = newLeft; 8298 mRightVolume = newRight; 8299 8300 // second get volume update from volume controller 8301 if (ctrlIdx >= 0) { 8302 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8303 mNewLeftVolume = newLeft; 8304 mNewRightVolume = newRight; 8305 } 8306 // then indicate volume to all other effects in chain. 8307 // Pass altered volume to effects before volume controller 8308 // and requested volume to effects after controller 8309 uint32_t lVol = newLeft; 8310 uint32_t rVol = newRight; 8311 8312 for (size_t i = 0; i < size; i++) { 8313 if ((int)i == ctrlIdx) continue; 8314 // this also works for ctrlIdx == -1 when there is no volume controller 8315 if ((int)i > ctrlIdx) { 8316 lVol = *left; 8317 rVol = *right; 8318 } 8319 mEffects[i]->setVolume(&lVol, &rVol, false); 8320 } 8321 *left = newLeft; 8322 *right = newRight; 8323 8324 return hasControl; 8325} 8326 8327status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8328{ 8329 const size_t SIZE = 256; 8330 char buffer[SIZE]; 8331 String8 result; 8332 8333 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8334 result.append(buffer); 8335 8336 bool locked = tryLock(mLock); 8337 // failed to lock - AudioFlinger is probably deadlocked 8338 if (!locked) { 8339 result.append("\tCould not lock mutex:\n"); 8340 } 8341 8342 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8343 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8344 mEffects.size(), 8345 (uint32_t)mInBuffer, 8346 (uint32_t)mOutBuffer, 8347 mActiveTrackCnt); 8348 result.append(buffer); 8349 write(fd, result.string(), result.size()); 8350 8351 for (size_t i = 0; i < mEffects.size(); ++i) { 8352 sp<EffectModule> effect = mEffects[i]; 8353 if (effect != 0) { 8354 effect->dump(fd, args); 8355 } 8356 } 8357 8358 if (locked) { 8359 mLock.unlock(); 8360 } 8361 8362 return NO_ERROR; 8363} 8364 8365// must be called with ThreadBase::mLock held 8366void AudioFlinger::EffectChain::setEffectSuspended_l( 8367 const effect_uuid_t *type, bool suspend) 8368{ 8369 sp<SuspendedEffectDesc> desc; 8370 // use effect type UUID timelow as key as there is no real risk of identical 8371 // timeLow fields among effect type UUIDs. 8372 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8373 if (suspend) { 8374 if (index >= 0) { 8375 desc = mSuspendedEffects.valueAt(index); 8376 } else { 8377 desc = new SuspendedEffectDesc(); 8378 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8379 mSuspendedEffects.add(type->timeLow, desc); 8380 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8381 } 8382 if (desc->mRefCount++ == 0) { 8383 sp<EffectModule> effect = getEffectIfEnabled(type); 8384 if (effect != 0) { 8385 desc->mEffect = effect; 8386 effect->setSuspended(true); 8387 effect->setEnabled(false); 8388 } 8389 } 8390 } else { 8391 if (index < 0) { 8392 return; 8393 } 8394 desc = mSuspendedEffects.valueAt(index); 8395 if (desc->mRefCount <= 0) { 8396 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8397 desc->mRefCount = 1; 8398 } 8399 if (--desc->mRefCount == 0) { 8400 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8401 if (desc->mEffect != 0) { 8402 sp<EffectModule> effect = desc->mEffect.promote(); 8403 if (effect != 0) { 8404 effect->setSuspended(false); 8405 sp<EffectHandle> handle = effect->controlHandle(); 8406 if (handle != 0) { 8407 effect->setEnabled(handle->enabled()); 8408 } 8409 } 8410 desc->mEffect.clear(); 8411 } 8412 mSuspendedEffects.removeItemsAt(index); 8413 } 8414 } 8415} 8416 8417// must be called with ThreadBase::mLock held 8418void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8419{ 8420 sp<SuspendedEffectDesc> desc; 8421 8422 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8423 if (suspend) { 8424 if (index >= 0) { 8425 desc = mSuspendedEffects.valueAt(index); 8426 } else { 8427 desc = new SuspendedEffectDesc(); 8428 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8429 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8430 } 8431 if (desc->mRefCount++ == 0) { 8432 Vector< sp<EffectModule> > effects; 8433 getSuspendEligibleEffects(effects); 8434 for (size_t i = 0; i < effects.size(); i++) { 8435 setEffectSuspended_l(&effects[i]->desc().type, true); 8436 } 8437 } 8438 } else { 8439 if (index < 0) { 8440 return; 8441 } 8442 desc = mSuspendedEffects.valueAt(index); 8443 if (desc->mRefCount <= 0) { 8444 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8445 desc->mRefCount = 1; 8446 } 8447 if (--desc->mRefCount == 0) { 8448 Vector<const effect_uuid_t *> types; 8449 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8450 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8451 continue; 8452 } 8453 types.add(&mSuspendedEffects.valueAt(i)->mType); 8454 } 8455 for (size_t i = 0; i < types.size(); i++) { 8456 setEffectSuspended_l(types[i], false); 8457 } 8458 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8459 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8460 } 8461 } 8462} 8463 8464 8465// The volume effect is used for automated tests only 8466#ifndef OPENSL_ES_H_ 8467static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8468 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8469const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8470#endif //OPENSL_ES_H_ 8471 8472bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8473{ 8474 // auxiliary effects and visualizer are never suspended on output mix 8475 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8476 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8477 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8478 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8479 return false; 8480 } 8481 return true; 8482} 8483 8484void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8485{ 8486 effects.clear(); 8487 for (size_t i = 0; i < mEffects.size(); i++) { 8488 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8489 effects.add(mEffects[i]); 8490 } 8491 } 8492} 8493 8494sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8495 const effect_uuid_t *type) 8496{ 8497 sp<EffectModule> effect = getEffectFromType_l(type); 8498 return effect != 0 && effect->isEnabled() ? effect : 0; 8499} 8500 8501void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8502 bool enabled) 8503{ 8504 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8505 if (enabled) { 8506 if (index < 0) { 8507 // if the effect is not suspend check if all effects are suspended 8508 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8509 if (index < 0) { 8510 return; 8511 } 8512 if (!isEffectEligibleForSuspend(effect->desc())) { 8513 return; 8514 } 8515 setEffectSuspended_l(&effect->desc().type, enabled); 8516 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8517 if (index < 0) { 8518 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8519 return; 8520 } 8521 } 8522 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8523 effect->desc().type.timeLow); 8524 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8525 // if effect is requested to suspended but was not yet enabled, supend it now. 8526 if (desc->mEffect == 0) { 8527 desc->mEffect = effect; 8528 effect->setEnabled(false); 8529 effect->setSuspended(true); 8530 } 8531 } else { 8532 if (index < 0) { 8533 return; 8534 } 8535 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8536 effect->desc().type.timeLow); 8537 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8538 desc->mEffect.clear(); 8539 effect->setSuspended(false); 8540 } 8541} 8542 8543#undef LOG_TAG 8544#define LOG_TAG "AudioFlinger" 8545 8546// ---------------------------------------------------------------------------- 8547 8548status_t AudioFlinger::onTransact( 8549 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8550{ 8551 return BnAudioFlinger::onTransact(code, data, reply, flags); 8552} 8553 8554}; // namespace android 8555