AudioFlinger.cpp revision 1a9ed11a472493cac7f6dfcbfac2064526a493ed
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#undef ADD_BATTERY_DATA 41 42#ifdef ADD_BATTERY_DATA 43#include <media/IMediaPlayerService.h> 44#include <media/IMediaDeathNotifier.h> 45#endif 46 47#include <private/media/AudioTrackShared.h> 48#include <private/media/AudioEffectShared.h> 49 50#include <system/audio.h> 51#include <hardware/audio.h> 52 53#include "AudioMixer.h" 54#include "AudioFlinger.h" 55#include "ServiceUtilities.h" 56 57#include <media/EffectsFactoryApi.h> 58#include <audio_effects/effect_visualizer.h> 59#include <audio_effects/effect_ns.h> 60#include <audio_effects/effect_aec.h> 61 62#include <audio_utils/primitives.h> 63 64#include <powermanager/PowerManager.h> 65 66// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72#include <common_time/cc_helper.h> 73#include <common_time/local_clock.h> 74 75// ---------------------------------------------------------------------------- 76 77 78namespace android { 79 80static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 81static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 82 83static const float MAX_GAIN = 4096.0f; 84static const uint32_t MAX_GAIN_INT = 0x1000; 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95static const int kDumpLockRetries = 50; 96static const int kDumpLockSleepUs = 20000; 97 98// don't warn about blocked writes or record buffer overflows more often than this 99static const nsecs_t kWarningThrottleNs = seconds(5); 100 101// RecordThread loop sleep time upon application overrun or audio HAL read error 102static const int kRecordThreadSleepUs = 5000; 103 104// maximum time to wait for setParameters to complete 105static const nsecs_t kSetParametersTimeoutNs = seconds(2); 106 107// minimum sleep time for the mixer thread loop when tracks are active but in underrun 108static const uint32_t kMinThreadSleepTimeUs = 5000; 109// maximum divider applied to the active sleep time in the mixer thread loop 110static const uint32_t kMaxThreadSleepTimeShift = 2; 111 112nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 113 114// ---------------------------------------------------------------------------- 115 116#ifdef ADD_BATTERY_DATA 117// To collect the amplifier usage 118static void addBatteryData(uint32_t params) { 119 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 120 if (service == NULL) { 121 // it already logged 122 return; 123 } 124 125 service->addBatteryData(params); 126} 127#endif 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152static const char * const audio_interfaces[] = { 153 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 154 AUDIO_HARDWARE_MODULE_ID_A2DP, 155 AUDIO_HARDWARE_MODULE_ID_USB, 156}; 157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 158 159// ---------------------------------------------------------------------------- 160 161AudioFlinger::AudioFlinger() 162 : BnAudioFlinger(), 163 mPrimaryHardwareDev(NULL), 164 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 165 mMasterVolume(1.0f), 166 mMasterVolumeSupportLvl(MVS_NONE), 167 mMasterMute(false), 168 mNextUniqueId(1), 169 mMode(AUDIO_MODE_INVALID), 170 mBtNrecIsOff(false) 171{ 172} 173 174void AudioFlinger::onFirstRef() 175{ 176 int rc = 0; 177 178 Mutex::Autolock _l(mLock); 179 180 /* TODO: move all this work into an Init() function */ 181 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 182 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 183 uint32_t int_val; 184 if (1 == sscanf(val_str, "%u", &int_val)) { 185 mStandbyTimeInNsecs = milliseconds(int_val); 186 ALOGI("Using %u mSec as standby time.", int_val); 187 } else { 188 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 189 ALOGI("Using default %u mSec as standby time.", 190 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 191 } 192 } 193 194 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 195 const hw_module_t *mod; 196 audio_hw_device_t *dev; 197 198 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 199 if (rc) 200 continue; 201 202 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 203 mod->name, mod->id); 204 mAudioHwDevs.push(dev); 205 206 if (mPrimaryHardwareDev == NULL) { 207 mPrimaryHardwareDev = dev; 208 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 209 mod->name, mod->id, audio_interfaces[i]); 210 } 211 } 212 213 if (mPrimaryHardwareDev == NULL) { 214 ALOGE("Primary audio interface not found"); 215 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 216 } 217 218 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 219 // primary HW dev is selected can change so these conditions might not always be equivalent. 220 // When that happens, re-visit all the code that assumes this. 221 222 AutoMutex lock(mHardwareLock); 223 224 // Determine the level of master volume support the primary audio HAL has, 225 // and set the initial master volume at the same time. 226 float initialVolume = 1.0; 227 mMasterVolumeSupportLvl = MVS_NONE; 228 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 229 audio_hw_device_t *dev = mPrimaryHardwareDev; 230 231 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 232 if ((NULL != dev->get_master_volume) && 233 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 234 mMasterVolumeSupportLvl = MVS_FULL; 235 } else { 236 mMasterVolumeSupportLvl = MVS_SETONLY; 237 initialVolume = 1.0; 238 } 239 240 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 241 if ((NULL == dev->set_master_volume) || 242 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 243 mMasterVolumeSupportLvl = MVS_NONE; 244 } 245 mHardwareStatus = AUDIO_HW_IDLE; 246 } 247 248 // Set the mode for each audio HAL, and try to set the initial volume (if 249 // supported) for all of the non-primary audio HALs. 250 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 251 audio_hw_device_t *dev = mAudioHwDevs[i]; 252 253 mHardwareStatus = AUDIO_HW_INIT; 254 rc = dev->init_check(dev); 255 mHardwareStatus = AUDIO_HW_IDLE; 256 if (rc == 0) { 257 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 258 mHardwareStatus = AUDIO_HW_SET_MODE; 259 dev->set_mode(dev, mMode); 260 261 if ((dev != mPrimaryHardwareDev) && 262 (NULL != dev->set_master_volume)) { 263 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 264 dev->set_master_volume(dev, initialVolume); 265 } 266 267 mHardwareStatus = AUDIO_HW_IDLE; 268 } 269 } 270 271 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 272 ? initialVolume 273 : 1.0; 274 mMasterVolume = initialVolume; 275 mHardwareStatus = AUDIO_HW_IDLE; 276} 277 278AudioFlinger::~AudioFlinger() 279{ 280 281 while (!mRecordThreads.isEmpty()) { 282 // closeInput() will remove first entry from mRecordThreads 283 closeInput(mRecordThreads.keyAt(0)); 284 } 285 while (!mPlaybackThreads.isEmpty()) { 286 // closeOutput() will remove first entry from mPlaybackThreads 287 closeOutput(mPlaybackThreads.keyAt(0)); 288 } 289 290 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 291 // no mHardwareLock needed, as there are no other references to this 292 audio_hw_device_close(mAudioHwDevs[i]); 293 } 294} 295 296audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 297{ 298 /* first matching HW device is returned */ 299 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 300 audio_hw_device_t *dev = mAudioHwDevs[i]; 301 if ((dev->get_supported_devices(dev) & devices) == devices) 302 return dev; 303 } 304 return NULL; 305} 306 307status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 308{ 309 const size_t SIZE = 256; 310 char buffer[SIZE]; 311 String8 result; 312 313 result.append("Clients:\n"); 314 for (size_t i = 0; i < mClients.size(); ++i) { 315 sp<Client> client = mClients.valueAt(i).promote(); 316 if (client != 0) { 317 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 318 result.append(buffer); 319 } 320 } 321 322 result.append("Global session refs:\n"); 323 result.append(" session pid count\n"); 324 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 325 AudioSessionRef *r = mAudioSessionRefs[i]; 326 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 327 result.append(buffer); 328 } 329 write(fd, result.string(), result.size()); 330 return NO_ERROR; 331} 332 333 334status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 335{ 336 const size_t SIZE = 256; 337 char buffer[SIZE]; 338 String8 result; 339 hardware_call_state hardwareStatus = mHardwareStatus; 340 341 snprintf(buffer, SIZE, "Hardware status: %d\n" 342 "Standby Time mSec: %u\n", 343 hardwareStatus, 344 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 345 result.append(buffer); 346 write(fd, result.string(), result.size()); 347 return NO_ERROR; 348} 349 350status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 351{ 352 const size_t SIZE = 256; 353 char buffer[SIZE]; 354 String8 result; 355 snprintf(buffer, SIZE, "Permission Denial: " 356 "can't dump AudioFlinger from pid=%d, uid=%d\n", 357 IPCThreadState::self()->getCallingPid(), 358 IPCThreadState::self()->getCallingUid()); 359 result.append(buffer); 360 write(fd, result.string(), result.size()); 361 return NO_ERROR; 362} 363 364static bool tryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = tryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = tryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 dumpClients(fd, args); 400 dumpInternals(fd, args); 401 402 // dump playback threads 403 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 404 mPlaybackThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump record threads 408 for (size_t i = 0; i < mRecordThreads.size(); i++) { 409 mRecordThreads.valueAt(i)->dump(fd, args); 410 } 411 412 // dump all hardware devs 413 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 414 audio_hw_device_t *dev = mAudioHwDevs[i]; 415 dev->dump(dev, fd); 416 } 417 if (locked) mLock.unlock(); 418 } 419 return NO_ERROR; 420} 421 422sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 423{ 424 // If pid is already in the mClients wp<> map, then use that entry 425 // (for which promote() is always != 0), otherwise create a new entry and Client. 426 sp<Client> client = mClients.valueFor(pid).promote(); 427 if (client == 0) { 428 client = new Client(this, pid); 429 mClients.add(pid, client); 430 } 431 432 return client; 433} 434 435// IAudioFlinger interface 436 437 438sp<IAudioTrack> AudioFlinger::createTrack( 439 pid_t pid, 440 audio_stream_type_t streamType, 441 uint32_t sampleRate, 442 audio_format_t format, 443 uint32_t channelMask, 444 int frameCount, 445 IAudioFlinger::track_flags_t flags, 446 const sp<IMemory>& sharedBuffer, 447 audio_io_handle_t output, 448 int *sessionId, 449 status_t *status) 450{ 451 sp<PlaybackThread::Track> track; 452 sp<TrackHandle> trackHandle; 453 sp<Client> client; 454 status_t lStatus; 455 int lSessionId; 456 457 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 458 // but if someone uses binder directly they could bypass that and cause us to crash 459 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 460 ALOGE("createTrack() invalid stream type %d", streamType); 461 lStatus = BAD_VALUE; 462 goto Exit; 463 } 464 465 { 466 Mutex::Autolock _l(mLock); 467 PlaybackThread *thread = checkPlaybackThread_l(output); 468 PlaybackThread *effectThread = NULL; 469 if (thread == NULL) { 470 ALOGE("unknown output thread"); 471 lStatus = BAD_VALUE; 472 goto Exit; 473 } 474 475 client = registerPid_l(pid); 476 477 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 478 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 479 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 480 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 481 if (mPlaybackThreads.keyAt(i) != output) { 482 // prevent same audio session on different output threads 483 uint32_t sessions = t->hasAudioSession(*sessionId); 484 if (sessions & PlaybackThread::TRACK_SESSION) { 485 ALOGE("createTrack() session ID %d already in use", *sessionId); 486 lStatus = BAD_VALUE; 487 goto Exit; 488 } 489 // check if an effect with same session ID is waiting for a track to be created 490 if (sessions & PlaybackThread::EFFECT_SESSION) { 491 effectThread = t.get(); 492 } 493 } 494 } 495 lSessionId = *sessionId; 496 } else { 497 // if no audio session id is provided, create one here 498 lSessionId = nextUniqueId(); 499 if (sessionId != NULL) { 500 *sessionId = lSessionId; 501 } 502 } 503 ALOGV("createTrack() lSessionId: %d", lSessionId); 504 505 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 506 track = thread->createTrack_l(client, streamType, sampleRate, format, 507 channelMask, frameCount, sharedBuffer, lSessionId, flags, &lStatus); 508 509 // move effect chain to this output thread if an effect on same session was waiting 510 // for a track to be created 511 if (lStatus == NO_ERROR && effectThread != NULL) { 512 Mutex::Autolock _dl(thread->mLock); 513 Mutex::Autolock _sl(effectThread->mLock); 514 moveEffectChain_l(lSessionId, effectThread, thread, true); 515 } 516 517 // Look for sync events awaiting for a session to be used. 518 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 519 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 520 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 521 track->setSyncEvent(mPendingSyncEvents[i]); 522 mPendingSyncEvents.removeAt(i); 523 i--; 524 } 525 } 526 } 527 } 528 if (lStatus == NO_ERROR) { 529 trackHandle = new TrackHandle(track); 530 } else { 531 // remove local strong reference to Client before deleting the Track so that the Client 532 // destructor is called by the TrackBase destructor with mLock held 533 client.clear(); 534 track.clear(); 535 } 536 537Exit: 538 if (status != NULL) { 539 *status = lStatus; 540 } 541 return trackHandle; 542} 543 544uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 545{ 546 Mutex::Autolock _l(mLock); 547 PlaybackThread *thread = checkPlaybackThread_l(output); 548 if (thread == NULL) { 549 ALOGW("sampleRate() unknown thread %d", output); 550 return 0; 551 } 552 return thread->sampleRate(); 553} 554 555int AudioFlinger::channelCount(audio_io_handle_t output) const 556{ 557 Mutex::Autolock _l(mLock); 558 PlaybackThread *thread = checkPlaybackThread_l(output); 559 if (thread == NULL) { 560 ALOGW("channelCount() unknown thread %d", output); 561 return 0; 562 } 563 return thread->channelCount(); 564} 565 566audio_format_t AudioFlinger::format(audio_io_handle_t output) const 567{ 568 Mutex::Autolock _l(mLock); 569 PlaybackThread *thread = checkPlaybackThread_l(output); 570 if (thread == NULL) { 571 ALOGW("format() unknown thread %d", output); 572 return AUDIO_FORMAT_INVALID; 573 } 574 return thread->format(); 575} 576 577size_t AudioFlinger::frameCount(audio_io_handle_t output) const 578{ 579 Mutex::Autolock _l(mLock); 580 PlaybackThread *thread = checkPlaybackThread_l(output); 581 if (thread == NULL) { 582 ALOGW("frameCount() unknown thread %d", output); 583 return 0; 584 } 585 return thread->frameCount(); 586} 587 588uint32_t AudioFlinger::latency(audio_io_handle_t output) const 589{ 590 Mutex::Autolock _l(mLock); 591 PlaybackThread *thread = checkPlaybackThread_l(output); 592 if (thread == NULL) { 593 ALOGW("latency() unknown thread %d", output); 594 return 0; 595 } 596 return thread->latency(); 597} 598 599status_t AudioFlinger::setMasterVolume(float value) 600{ 601 status_t ret = initCheck(); 602 if (ret != NO_ERROR) { 603 return ret; 604 } 605 606 // check calling permissions 607 if (!settingsAllowed()) { 608 return PERMISSION_DENIED; 609 } 610 611 float swmv = value; 612 613 // when hw supports master volume, don't scale in sw mixer 614 if (MVS_NONE != mMasterVolumeSupportLvl) { 615 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 616 AutoMutex lock(mHardwareLock); 617 audio_hw_device_t *dev = mAudioHwDevs[i]; 618 619 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 620 if (NULL != dev->set_master_volume) { 621 dev->set_master_volume(dev, value); 622 } 623 mHardwareStatus = AUDIO_HW_IDLE; 624 } 625 626 swmv = 1.0; 627 } 628 629 Mutex::Autolock _l(mLock); 630 mMasterVolume = value; 631 mMasterVolumeSW = swmv; 632 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 633 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 634 635 return NO_ERROR; 636} 637 638status_t AudioFlinger::setMode(audio_mode_t mode) 639{ 640 status_t ret = initCheck(); 641 if (ret != NO_ERROR) { 642 return ret; 643 } 644 645 // check calling permissions 646 if (!settingsAllowed()) { 647 return PERMISSION_DENIED; 648 } 649 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 650 ALOGW("Illegal value: setMode(%d)", mode); 651 return BAD_VALUE; 652 } 653 654 { // scope for the lock 655 AutoMutex lock(mHardwareLock); 656 mHardwareStatus = AUDIO_HW_SET_MODE; 657 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 658 mHardwareStatus = AUDIO_HW_IDLE; 659 } 660 661 if (NO_ERROR == ret) { 662 Mutex::Autolock _l(mLock); 663 mMode = mode; 664 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 665 mPlaybackThreads.valueAt(i)->setMode(mode); 666 } 667 668 return ret; 669} 670 671status_t AudioFlinger::setMicMute(bool state) 672{ 673 status_t ret = initCheck(); 674 if (ret != NO_ERROR) { 675 return ret; 676 } 677 678 // check calling permissions 679 if (!settingsAllowed()) { 680 return PERMISSION_DENIED; 681 } 682 683 AutoMutex lock(mHardwareLock); 684 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 685 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 686 mHardwareStatus = AUDIO_HW_IDLE; 687 return ret; 688} 689 690bool AudioFlinger::getMicMute() const 691{ 692 status_t ret = initCheck(); 693 if (ret != NO_ERROR) { 694 return false; 695 } 696 697 bool state = AUDIO_MODE_INVALID; 698 AutoMutex lock(mHardwareLock); 699 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 700 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 701 mHardwareStatus = AUDIO_HW_IDLE; 702 return state; 703} 704 705status_t AudioFlinger::setMasterMute(bool muted) 706{ 707 // check calling permissions 708 if (!settingsAllowed()) { 709 return PERMISSION_DENIED; 710 } 711 712 Mutex::Autolock _l(mLock); 713 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 714 mMasterMute = muted; 715 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 716 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 717 718 return NO_ERROR; 719} 720 721float AudioFlinger::masterVolume() const 722{ 723 Mutex::Autolock _l(mLock); 724 return masterVolume_l(); 725} 726 727float AudioFlinger::masterVolumeSW() const 728{ 729 Mutex::Autolock _l(mLock); 730 return masterVolumeSW_l(); 731} 732 733bool AudioFlinger::masterMute() const 734{ 735 Mutex::Autolock _l(mLock); 736 return masterMute_l(); 737} 738 739float AudioFlinger::masterVolume_l() const 740{ 741 if (MVS_FULL == mMasterVolumeSupportLvl) { 742 float ret_val; 743 AutoMutex lock(mHardwareLock); 744 745 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 746 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 747 (NULL != mPrimaryHardwareDev->get_master_volume), 748 "can't get master volume"); 749 750 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 751 mHardwareStatus = AUDIO_HW_IDLE; 752 return ret_val; 753 } 754 755 return mMasterVolume; 756} 757 758status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 759 audio_io_handle_t output) 760{ 761 // check calling permissions 762 if (!settingsAllowed()) { 763 return PERMISSION_DENIED; 764 } 765 766 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 767 ALOGE("setStreamVolume() invalid stream %d", stream); 768 return BAD_VALUE; 769 } 770 771 AutoMutex lock(mLock); 772 PlaybackThread *thread = NULL; 773 if (output) { 774 thread = checkPlaybackThread_l(output); 775 if (thread == NULL) { 776 return BAD_VALUE; 777 } 778 } 779 780 mStreamTypes[stream].volume = value; 781 782 if (thread == NULL) { 783 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 784 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 785 } 786 } else { 787 thread->setStreamVolume(stream, value); 788 } 789 790 return NO_ERROR; 791} 792 793status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 794{ 795 // check calling permissions 796 if (!settingsAllowed()) { 797 return PERMISSION_DENIED; 798 } 799 800 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 801 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 802 ALOGE("setStreamMute() invalid stream %d", stream); 803 return BAD_VALUE; 804 } 805 806 AutoMutex lock(mLock); 807 mStreamTypes[stream].mute = muted; 808 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 809 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 810 811 return NO_ERROR; 812} 813 814float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return 0.0f; 818 } 819 820 AutoMutex lock(mLock); 821 float volume; 822 if (output) { 823 PlaybackThread *thread = checkPlaybackThread_l(output); 824 if (thread == NULL) { 825 return 0.0f; 826 } 827 volume = thread->streamVolume(stream); 828 } else { 829 volume = streamVolume_l(stream); 830 } 831 832 return volume; 833} 834 835bool AudioFlinger::streamMute(audio_stream_type_t stream) const 836{ 837 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 838 return true; 839 } 840 841 AutoMutex lock(mLock); 842 return streamMute_l(stream); 843} 844 845status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 846{ 847 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 848 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 849 // check calling permissions 850 if (!settingsAllowed()) { 851 return PERMISSION_DENIED; 852 } 853 854 // ioHandle == 0 means the parameters are global to the audio hardware interface 855 if (ioHandle == 0) { 856 status_t final_result = NO_ERROR; 857 { 858 AutoMutex lock(mHardwareLock); 859 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 860 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 861 audio_hw_device_t *dev = mAudioHwDevs[i]; 862 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 863 final_result = result ?: final_result; 864 } 865 mHardwareStatus = AUDIO_HW_IDLE; 866 } 867 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 868 AudioParameter param = AudioParameter(keyValuePairs); 869 String8 value; 870 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 871 Mutex::Autolock _l(mLock); 872 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 873 if (mBtNrecIsOff != btNrecIsOff) { 874 for (size_t i = 0; i < mRecordThreads.size(); i++) { 875 sp<RecordThread> thread = mRecordThreads.valueAt(i); 876 RecordThread::RecordTrack *track = thread->track(); 877 if (track != NULL) { 878 audio_devices_t device = (audio_devices_t)( 879 thread->device() & AUDIO_DEVICE_IN_ALL); 880 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 881 thread->setEffectSuspended(FX_IID_AEC, 882 suspend, 883 track->sessionId()); 884 thread->setEffectSuspended(FX_IID_NS, 885 suspend, 886 track->sessionId()); 887 } 888 } 889 mBtNrecIsOff = btNrecIsOff; 890 } 891 } 892 return final_result; 893 } 894 895 // hold a strong ref on thread in case closeOutput() or closeInput() is called 896 // and the thread is exited once the lock is released 897 sp<ThreadBase> thread; 898 { 899 Mutex::Autolock _l(mLock); 900 thread = checkPlaybackThread_l(ioHandle); 901 if (thread == NULL) { 902 thread = checkRecordThread_l(ioHandle); 903 } else if (thread == primaryPlaybackThread_l()) { 904 // indicate output device change to all input threads for pre processing 905 AudioParameter param = AudioParameter(keyValuePairs); 906 int value; 907 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 908 (value != 0)) { 909 for (size_t i = 0; i < mRecordThreads.size(); i++) { 910 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 911 } 912 } 913 } 914 } 915 if (thread != 0) { 916 return thread->setParameters(keyValuePairs); 917 } 918 return BAD_VALUE; 919} 920 921String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 922{ 923// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 924// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 925 926 if (ioHandle == 0) { 927 String8 out_s8; 928 929 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 930 char *s; 931 { 932 AutoMutex lock(mHardwareLock); 933 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 934 audio_hw_device_t *dev = mAudioHwDevs[i]; 935 s = dev->get_parameters(dev, keys.string()); 936 mHardwareStatus = AUDIO_HW_IDLE; 937 } 938 out_s8 += String8(s ? s : ""); 939 free(s); 940 } 941 return out_s8; 942 } 943 944 Mutex::Autolock _l(mLock); 945 946 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 947 if (playbackThread != NULL) { 948 return playbackThread->getParameters(keys); 949 } 950 RecordThread *recordThread = checkRecordThread_l(ioHandle); 951 if (recordThread != NULL) { 952 return recordThread->getParameters(keys); 953 } 954 return String8(""); 955} 956 957size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 958{ 959 status_t ret = initCheck(); 960 if (ret != NO_ERROR) { 961 return 0; 962 } 963 964 AutoMutex lock(mHardwareLock); 965 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 966 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 967 mHardwareStatus = AUDIO_HW_IDLE; 968 return size; 969} 970 971unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 972{ 973 if (ioHandle == 0) { 974 return 0; 975 } 976 977 Mutex::Autolock _l(mLock); 978 979 RecordThread *recordThread = checkRecordThread_l(ioHandle); 980 if (recordThread != NULL) { 981 return recordThread->getInputFramesLost(); 982 } 983 return 0; 984} 985 986status_t AudioFlinger::setVoiceVolume(float value) 987{ 988 status_t ret = initCheck(); 989 if (ret != NO_ERROR) { 990 return ret; 991 } 992 993 // check calling permissions 994 if (!settingsAllowed()) { 995 return PERMISSION_DENIED; 996 } 997 998 AutoMutex lock(mHardwareLock); 999 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1000 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1001 mHardwareStatus = AUDIO_HW_IDLE; 1002 1003 return ret; 1004} 1005 1006status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1007 audio_io_handle_t output) const 1008{ 1009 status_t status; 1010 1011 Mutex::Autolock _l(mLock); 1012 1013 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1014 if (playbackThread != NULL) { 1015 return playbackThread->getRenderPosition(halFrames, dspFrames); 1016 } 1017 1018 return BAD_VALUE; 1019} 1020 1021void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1022{ 1023 1024 Mutex::Autolock _l(mLock); 1025 1026 pid_t pid = IPCThreadState::self()->getCallingPid(); 1027 if (mNotificationClients.indexOfKey(pid) < 0) { 1028 sp<NotificationClient> notificationClient = new NotificationClient(this, 1029 client, 1030 pid); 1031 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1032 1033 mNotificationClients.add(pid, notificationClient); 1034 1035 sp<IBinder> binder = client->asBinder(); 1036 binder->linkToDeath(notificationClient); 1037 1038 // the config change is always sent from playback or record threads to avoid deadlock 1039 // with AudioSystem::gLock 1040 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1041 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1042 } 1043 1044 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1045 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1046 } 1047 } 1048} 1049 1050void AudioFlinger::removeNotificationClient(pid_t pid) 1051{ 1052 Mutex::Autolock _l(mLock); 1053 1054 mNotificationClients.removeItem(pid); 1055 1056 ALOGV("%d died, releasing its sessions", pid); 1057 size_t num = mAudioSessionRefs.size(); 1058 bool removed = false; 1059 for (size_t i = 0; i< num; ) { 1060 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1061 ALOGV(" pid %d @ %d", ref->mPid, i); 1062 if (ref->mPid == pid) { 1063 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1064 mAudioSessionRefs.removeAt(i); 1065 delete ref; 1066 removed = true; 1067 num--; 1068 } else { 1069 i++; 1070 } 1071 } 1072 if (removed) { 1073 purgeStaleEffects_l(); 1074 } 1075} 1076 1077// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1078void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1079{ 1080 size_t size = mNotificationClients.size(); 1081 for (size_t i = 0; i < size; i++) { 1082 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1083 param2); 1084 } 1085} 1086 1087// removeClient_l() must be called with AudioFlinger::mLock held 1088void AudioFlinger::removeClient_l(pid_t pid) 1089{ 1090 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1091 mClients.removeItem(pid); 1092} 1093 1094 1095// ---------------------------------------------------------------------------- 1096 1097AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1098 uint32_t device, type_t type) 1099 : Thread(false), 1100 mType(type), 1101 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1102 // mChannelMask 1103 mChannelCount(0), 1104 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1105 mParamStatus(NO_ERROR), 1106 mStandby(false), mId(id), 1107 mDevice(device), 1108 mDeathRecipient(new PMDeathRecipient(this)) 1109{ 1110} 1111 1112AudioFlinger::ThreadBase::~ThreadBase() 1113{ 1114 mParamCond.broadcast(); 1115 // do not lock the mutex in destructor 1116 releaseWakeLock_l(); 1117 if (mPowerManager != 0) { 1118 sp<IBinder> binder = mPowerManager->asBinder(); 1119 binder->unlinkToDeath(mDeathRecipient); 1120 } 1121} 1122 1123void AudioFlinger::ThreadBase::exit() 1124{ 1125 ALOGV("ThreadBase::exit"); 1126 { 1127 // This lock prevents the following race in thread (uniprocessor for illustration): 1128 // if (!exitPending()) { 1129 // // context switch from here to exit() 1130 // // exit() calls requestExit(), what exitPending() observes 1131 // // exit() calls signal(), which is dropped since no waiters 1132 // // context switch back from exit() to here 1133 // mWaitWorkCV.wait(...); 1134 // // now thread is hung 1135 // } 1136 AutoMutex lock(mLock); 1137 requestExit(); 1138 mWaitWorkCV.signal(); 1139 } 1140 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1141 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1142 requestExitAndWait(); 1143} 1144 1145status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1146{ 1147 status_t status; 1148 1149 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1150 Mutex::Autolock _l(mLock); 1151 1152 mNewParameters.add(keyValuePairs); 1153 mWaitWorkCV.signal(); 1154 // wait condition with timeout in case the thread loop has exited 1155 // before the request could be processed 1156 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1157 status = mParamStatus; 1158 mWaitWorkCV.signal(); 1159 } else { 1160 status = TIMED_OUT; 1161 } 1162 return status; 1163} 1164 1165void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1166{ 1167 Mutex::Autolock _l(mLock); 1168 sendConfigEvent_l(event, param); 1169} 1170 1171// sendConfigEvent_l() must be called with ThreadBase::mLock held 1172void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1173{ 1174 ConfigEvent configEvent; 1175 configEvent.mEvent = event; 1176 configEvent.mParam = param; 1177 mConfigEvents.add(configEvent); 1178 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1179 mWaitWorkCV.signal(); 1180} 1181 1182void AudioFlinger::ThreadBase::processConfigEvents() 1183{ 1184 mLock.lock(); 1185 while (!mConfigEvents.isEmpty()) { 1186 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1187 ConfigEvent configEvent = mConfigEvents[0]; 1188 mConfigEvents.removeAt(0); 1189 // release mLock before locking AudioFlinger mLock: lock order is always 1190 // AudioFlinger then ThreadBase to avoid cross deadlock 1191 mLock.unlock(); 1192 mAudioFlinger->mLock.lock(); 1193 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1194 mAudioFlinger->mLock.unlock(); 1195 mLock.lock(); 1196 } 1197 mLock.unlock(); 1198} 1199 1200status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1201{ 1202 const size_t SIZE = 256; 1203 char buffer[SIZE]; 1204 String8 result; 1205 1206 bool locked = tryLock(mLock); 1207 if (!locked) { 1208 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1209 write(fd, buffer, strlen(buffer)); 1210 } 1211 1212 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1213 result.append(buffer); 1214 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1217 result.append(buffer); 1218 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1219 result.append(buffer); 1220 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1221 result.append(buffer); 1222 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1223 result.append(buffer); 1224 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1225 result.append(buffer); 1226 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1227 result.append(buffer); 1228 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1229 result.append(buffer); 1230 1231 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1232 result.append(buffer); 1233 result.append(" Index Command"); 1234 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1235 snprintf(buffer, SIZE, "\n %02d ", i); 1236 result.append(buffer); 1237 result.append(mNewParameters[i]); 1238 } 1239 1240 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1241 result.append(buffer); 1242 snprintf(buffer, SIZE, " Index event param\n"); 1243 result.append(buffer); 1244 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1245 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1246 result.append(buffer); 1247 } 1248 result.append("\n"); 1249 1250 write(fd, result.string(), result.size()); 1251 1252 if (locked) { 1253 mLock.unlock(); 1254 } 1255 return NO_ERROR; 1256} 1257 1258status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1259{ 1260 const size_t SIZE = 256; 1261 char buffer[SIZE]; 1262 String8 result; 1263 1264 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1265 write(fd, buffer, strlen(buffer)); 1266 1267 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1268 sp<EffectChain> chain = mEffectChains[i]; 1269 if (chain != 0) { 1270 chain->dump(fd, args); 1271 } 1272 } 1273 return NO_ERROR; 1274} 1275 1276void AudioFlinger::ThreadBase::acquireWakeLock() 1277{ 1278 Mutex::Autolock _l(mLock); 1279 acquireWakeLock_l(); 1280} 1281 1282void AudioFlinger::ThreadBase::acquireWakeLock_l() 1283{ 1284 if (mPowerManager == 0) { 1285 // use checkService() to avoid blocking if power service is not up yet 1286 sp<IBinder> binder = 1287 defaultServiceManager()->checkService(String16("power")); 1288 if (binder == 0) { 1289 ALOGW("Thread %s cannot connect to the power manager service", mName); 1290 } else { 1291 mPowerManager = interface_cast<IPowerManager>(binder); 1292 binder->linkToDeath(mDeathRecipient); 1293 } 1294 } 1295 if (mPowerManager != 0) { 1296 sp<IBinder> binder = new BBinder(); 1297 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1298 binder, 1299 String16(mName)); 1300 if (status == NO_ERROR) { 1301 mWakeLockToken = binder; 1302 } 1303 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1304 } 1305} 1306 1307void AudioFlinger::ThreadBase::releaseWakeLock() 1308{ 1309 Mutex::Autolock _l(mLock); 1310 releaseWakeLock_l(); 1311} 1312 1313void AudioFlinger::ThreadBase::releaseWakeLock_l() 1314{ 1315 if (mWakeLockToken != 0) { 1316 ALOGV("releaseWakeLock_l() %s", mName); 1317 if (mPowerManager != 0) { 1318 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1319 } 1320 mWakeLockToken.clear(); 1321 } 1322} 1323 1324void AudioFlinger::ThreadBase::clearPowerManager() 1325{ 1326 Mutex::Autolock _l(mLock); 1327 releaseWakeLock_l(); 1328 mPowerManager.clear(); 1329} 1330 1331void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1332{ 1333 sp<ThreadBase> thread = mThread.promote(); 1334 if (thread != 0) { 1335 thread->clearPowerManager(); 1336 } 1337 ALOGW("power manager service died !!!"); 1338} 1339 1340void AudioFlinger::ThreadBase::setEffectSuspended( 1341 const effect_uuid_t *type, bool suspend, int sessionId) 1342{ 1343 Mutex::Autolock _l(mLock); 1344 setEffectSuspended_l(type, suspend, sessionId); 1345} 1346 1347void AudioFlinger::ThreadBase::setEffectSuspended_l( 1348 const effect_uuid_t *type, bool suspend, int sessionId) 1349{ 1350 sp<EffectChain> chain = getEffectChain_l(sessionId); 1351 if (chain != 0) { 1352 if (type != NULL) { 1353 chain->setEffectSuspended_l(type, suspend); 1354 } else { 1355 chain->setEffectSuspendedAll_l(suspend); 1356 } 1357 } 1358 1359 updateSuspendedSessions_l(type, suspend, sessionId); 1360} 1361 1362void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1363{ 1364 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1365 if (index < 0) { 1366 return; 1367 } 1368 1369 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1370 mSuspendedSessions.editValueAt(index); 1371 1372 for (size_t i = 0; i < sessionEffects.size(); i++) { 1373 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1374 for (int j = 0; j < desc->mRefCount; j++) { 1375 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1376 chain->setEffectSuspendedAll_l(true); 1377 } else { 1378 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1379 desc->mType.timeLow); 1380 chain->setEffectSuspended_l(&desc->mType, true); 1381 } 1382 } 1383 } 1384} 1385 1386void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1387 bool suspend, 1388 int sessionId) 1389{ 1390 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1391 1392 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1393 1394 if (suspend) { 1395 if (index >= 0) { 1396 sessionEffects = mSuspendedSessions.editValueAt(index); 1397 } else { 1398 mSuspendedSessions.add(sessionId, sessionEffects); 1399 } 1400 } else { 1401 if (index < 0) { 1402 return; 1403 } 1404 sessionEffects = mSuspendedSessions.editValueAt(index); 1405 } 1406 1407 1408 int key = EffectChain::kKeyForSuspendAll; 1409 if (type != NULL) { 1410 key = type->timeLow; 1411 } 1412 index = sessionEffects.indexOfKey(key); 1413 1414 sp<SuspendedSessionDesc> desc; 1415 if (suspend) { 1416 if (index >= 0) { 1417 desc = sessionEffects.valueAt(index); 1418 } else { 1419 desc = new SuspendedSessionDesc(); 1420 if (type != NULL) { 1421 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1422 } 1423 sessionEffects.add(key, desc); 1424 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1425 } 1426 desc->mRefCount++; 1427 } else { 1428 if (index < 0) { 1429 return; 1430 } 1431 desc = sessionEffects.valueAt(index); 1432 if (--desc->mRefCount == 0) { 1433 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1434 sessionEffects.removeItemsAt(index); 1435 if (sessionEffects.isEmpty()) { 1436 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1437 sessionId); 1438 mSuspendedSessions.removeItem(sessionId); 1439 } 1440 } 1441 } 1442 if (!sessionEffects.isEmpty()) { 1443 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1444 } 1445} 1446 1447void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1448 bool enabled, 1449 int sessionId) 1450{ 1451 Mutex::Autolock _l(mLock); 1452 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1453} 1454 1455void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1456 bool enabled, 1457 int sessionId) 1458{ 1459 if (mType != RECORD) { 1460 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1461 // another session. This gives the priority to well behaved effect control panels 1462 // and applications not using global effects. 1463 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1464 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1465 } 1466 } 1467 1468 sp<EffectChain> chain = getEffectChain_l(sessionId); 1469 if (chain != 0) { 1470 chain->checkSuspendOnEffectEnabled(effect, enabled); 1471 } 1472} 1473 1474// ---------------------------------------------------------------------------- 1475 1476AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1477 AudioStreamOut* output, 1478 audio_io_handle_t id, 1479 uint32_t device, 1480 type_t type) 1481 : ThreadBase(audioFlinger, id, device, type), 1482 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1483 // Assumes constructor is called by AudioFlinger with it's mLock held, 1484 // but it would be safer to explicitly pass initial masterMute as parameter 1485 mMasterMute(audioFlinger->masterMute_l()), 1486 // mStreamTypes[] initialized in constructor body 1487 mOutput(output), 1488 // Assumes constructor is called by AudioFlinger with it's mLock held, 1489 // but it would be safer to explicitly pass initial masterVolume as parameter 1490 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1491 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1492 mMixerStatus(MIXER_IDLE), 1493 mPrevMixerStatus(MIXER_IDLE), 1494 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1495{ 1496 snprintf(mName, kNameLength, "AudioOut_%X", id); 1497 1498 readOutputParameters(); 1499 1500 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1501 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1502 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1503 stream = (audio_stream_type_t) (stream + 1)) { 1504 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1505 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1506 } 1507 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1508 // because mAudioFlinger doesn't have one to copy from 1509} 1510 1511AudioFlinger::PlaybackThread::~PlaybackThread() 1512{ 1513 delete [] mMixBuffer; 1514} 1515 1516status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1517{ 1518 dumpInternals(fd, args); 1519 dumpTracks(fd, args); 1520 dumpEffectChains(fd, args); 1521 return NO_ERROR; 1522} 1523 1524status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1525{ 1526 const size_t SIZE = 256; 1527 char buffer[SIZE]; 1528 String8 result; 1529 1530 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1531 result.append(buffer); 1532 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1533 for (size_t i = 0; i < mTracks.size(); ++i) { 1534 sp<Track> track = mTracks[i]; 1535 if (track != 0) { 1536 track->dump(buffer, SIZE); 1537 result.append(buffer); 1538 } 1539 } 1540 1541 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1542 result.append(buffer); 1543 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1544 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1545 sp<Track> track = mActiveTracks[i].promote(); 1546 if (track != 0) { 1547 track->dump(buffer, SIZE); 1548 result.append(buffer); 1549 } 1550 } 1551 write(fd, result.string(), result.size()); 1552 return NO_ERROR; 1553} 1554 1555status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1556{ 1557 const size_t SIZE = 256; 1558 char buffer[SIZE]; 1559 String8 result; 1560 1561 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1562 result.append(buffer); 1563 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1564 result.append(buffer); 1565 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1566 result.append(buffer); 1567 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1568 result.append(buffer); 1569 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1570 result.append(buffer); 1571 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1572 result.append(buffer); 1573 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1574 result.append(buffer); 1575 write(fd, result.string(), result.size()); 1576 1577 dumpBase(fd, args); 1578 1579 return NO_ERROR; 1580} 1581 1582// Thread virtuals 1583status_t AudioFlinger::PlaybackThread::readyToRun() 1584{ 1585 status_t status = initCheck(); 1586 if (status == NO_ERROR) { 1587 ALOGI("AudioFlinger's thread %p ready to run", this); 1588 } else { 1589 ALOGE("No working audio driver found."); 1590 } 1591 return status; 1592} 1593 1594void AudioFlinger::PlaybackThread::onFirstRef() 1595{ 1596 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1597} 1598 1599// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1600sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1601 const sp<AudioFlinger::Client>& client, 1602 audio_stream_type_t streamType, 1603 uint32_t sampleRate, 1604 audio_format_t format, 1605 uint32_t channelMask, 1606 int frameCount, 1607 const sp<IMemory>& sharedBuffer, 1608 int sessionId, 1609 IAudioFlinger::track_flags_t flags, 1610 status_t *status) 1611{ 1612 sp<Track> track; 1613 status_t lStatus; 1614 1615 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1616 1617 // client expresses a preference for FAST, but we get the final say 1618 if ((flags & IAudioFlinger::TRACK_FAST) && 1619 !( 1620 // not timed 1621 (!isTimed) && 1622 // either of these use cases: 1623 ( 1624 // use case 1: shared buffer with any frame count 1625 ( 1626 (sharedBuffer != 0) 1627 ) || 1628 // use case 2: callback handler and small power-of-2 frame count 1629 ( 1630 // unfortunately we can't verify that there's a callback until start() 1631 // FIXME supported frame counts should not be hard-coded 1632 ( 1633 (frameCount == 128) || 1634 (frameCount == 256) || 1635 (frameCount == 512) 1636 ) 1637 ) 1638 ) && 1639 // PCM data 1640 audio_is_linear_pcm(format) && 1641 // mono or stereo 1642 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1643 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1644 // hardware sample rate 1645 (sampleRate == mSampleRate) 1646 // FIXME test that MixerThread for this fast track has a capable output HAL 1647 // FIXME add a permission test also? 1648 ) ) { 1649 ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied"); 1650 flags &= ~IAudioFlinger::TRACK_FAST; 1651 } 1652 1653 if (mType == DIRECT) { 1654 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1655 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1656 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1657 "for output %p with format %d", 1658 sampleRate, format, channelMask, mOutput, mFormat); 1659 lStatus = BAD_VALUE; 1660 goto Exit; 1661 } 1662 } 1663 } else { 1664 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1665 if (sampleRate > mSampleRate*2) { 1666 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1667 lStatus = BAD_VALUE; 1668 goto Exit; 1669 } 1670 } 1671 1672 lStatus = initCheck(); 1673 if (lStatus != NO_ERROR) { 1674 ALOGE("Audio driver not initialized."); 1675 goto Exit; 1676 } 1677 1678 { // scope for mLock 1679 Mutex::Autolock _l(mLock); 1680 1681 // all tracks in same audio session must share the same routing strategy otherwise 1682 // conflicts will happen when tracks are moved from one output to another by audio policy 1683 // manager 1684 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1685 for (size_t i = 0; i < mTracks.size(); ++i) { 1686 sp<Track> t = mTracks[i]; 1687 if (t != 0 && !t->isOutputTrack()) { 1688 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1689 if (sessionId == t->sessionId() && strategy != actual) { 1690 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1691 strategy, actual); 1692 lStatus = BAD_VALUE; 1693 goto Exit; 1694 } 1695 } 1696 } 1697 1698 if (!isTimed) { 1699 track = new Track(this, client, streamType, sampleRate, format, 1700 channelMask, frameCount, sharedBuffer, sessionId, flags); 1701 } else { 1702 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1703 channelMask, frameCount, sharedBuffer, sessionId); 1704 } 1705 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1706 lStatus = NO_MEMORY; 1707 goto Exit; 1708 } 1709 mTracks.add(track); 1710 1711 sp<EffectChain> chain = getEffectChain_l(sessionId); 1712 if (chain != 0) { 1713 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1714 track->setMainBuffer(chain->inBuffer()); 1715 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1716 chain->incTrackCnt(); 1717 } 1718 } 1719 lStatus = NO_ERROR; 1720 1721Exit: 1722 if (status) { 1723 *status = lStatus; 1724 } 1725 return track; 1726} 1727 1728uint32_t AudioFlinger::PlaybackThread::latency() const 1729{ 1730 Mutex::Autolock _l(mLock); 1731 if (initCheck() == NO_ERROR) { 1732 return mOutput->stream->get_latency(mOutput->stream); 1733 } else { 1734 return 0; 1735 } 1736} 1737 1738void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1739{ 1740 Mutex::Autolock _l(mLock); 1741 mMasterVolume = value; 1742} 1743 1744void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1745{ 1746 Mutex::Autolock _l(mLock); 1747 setMasterMute_l(muted); 1748} 1749 1750void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1751{ 1752 Mutex::Autolock _l(mLock); 1753 mStreamTypes[stream].volume = value; 1754} 1755 1756void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1757{ 1758 Mutex::Autolock _l(mLock); 1759 mStreamTypes[stream].mute = muted; 1760} 1761 1762float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1763{ 1764 Mutex::Autolock _l(mLock); 1765 return mStreamTypes[stream].volume; 1766} 1767 1768// addTrack_l() must be called with ThreadBase::mLock held 1769status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1770{ 1771 status_t status = ALREADY_EXISTS; 1772 1773 // set retry count for buffer fill 1774 track->mRetryCount = kMaxTrackStartupRetries; 1775 if (mActiveTracks.indexOf(track) < 0) { 1776 // the track is newly added, make sure it fills up all its 1777 // buffers before playing. This is to ensure the client will 1778 // effectively get the latency it requested. 1779 track->mFillingUpStatus = Track::FS_FILLING; 1780 track->mResetDone = false; 1781 mActiveTracks.add(track); 1782 if (track->mainBuffer() != mMixBuffer) { 1783 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1784 if (chain != 0) { 1785 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1786 chain->incActiveTrackCnt(); 1787 } 1788 } 1789 1790 status = NO_ERROR; 1791 } 1792 1793 ALOGV("mWaitWorkCV.broadcast"); 1794 mWaitWorkCV.broadcast(); 1795 1796 return status; 1797} 1798 1799// destroyTrack_l() must be called with ThreadBase::mLock held 1800void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1801{ 1802 track->mState = TrackBase::TERMINATED; 1803 if (mActiveTracks.indexOf(track) < 0) { 1804 removeTrack_l(track); 1805 } 1806} 1807 1808void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1809{ 1810 mTracks.remove(track); 1811 deleteTrackName_l(track->name()); 1812 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1813 if (chain != 0) { 1814 chain->decTrackCnt(); 1815 } 1816} 1817 1818String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1819{ 1820 String8 out_s8 = String8(""); 1821 char *s; 1822 1823 Mutex::Autolock _l(mLock); 1824 if (initCheck() != NO_ERROR) { 1825 return out_s8; 1826 } 1827 1828 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1829 out_s8 = String8(s); 1830 free(s); 1831 return out_s8; 1832} 1833 1834// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1835void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1836 AudioSystem::OutputDescriptor desc; 1837 void *param2 = NULL; 1838 1839 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1840 1841 switch (event) { 1842 case AudioSystem::OUTPUT_OPENED: 1843 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1844 desc.channels = mChannelMask; 1845 desc.samplingRate = mSampleRate; 1846 desc.format = mFormat; 1847 desc.frameCount = mFrameCount; 1848 desc.latency = latency(); 1849 param2 = &desc; 1850 break; 1851 1852 case AudioSystem::STREAM_CONFIG_CHANGED: 1853 param2 = ¶m; 1854 case AudioSystem::OUTPUT_CLOSED: 1855 default: 1856 break; 1857 } 1858 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1859} 1860 1861void AudioFlinger::PlaybackThread::readOutputParameters() 1862{ 1863 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1864 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1865 mChannelCount = (uint16_t)popcount(mChannelMask); 1866 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1867 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1868 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1869 1870 // FIXME - Current mixer implementation only supports stereo output: Always 1871 // Allocate a stereo buffer even if HW output is mono. 1872 delete[] mMixBuffer; 1873 mMixBuffer = new int16_t[mFrameCount * 2]; 1874 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1875 1876 // force reconfiguration of effect chains and engines to take new buffer size and audio 1877 // parameters into account 1878 // Note that mLock is not held when readOutputParameters() is called from the constructor 1879 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1880 // matter. 1881 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1882 Vector< sp<EffectChain> > effectChains = mEffectChains; 1883 for (size_t i = 0; i < effectChains.size(); i ++) { 1884 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1885 } 1886} 1887 1888status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1889{ 1890 if (halFrames == NULL || dspFrames == NULL) { 1891 return BAD_VALUE; 1892 } 1893 Mutex::Autolock _l(mLock); 1894 if (initCheck() != NO_ERROR) { 1895 return INVALID_OPERATION; 1896 } 1897 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1898 1899 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1900} 1901 1902uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1903{ 1904 Mutex::Autolock _l(mLock); 1905 uint32_t result = 0; 1906 if (getEffectChain_l(sessionId) != 0) { 1907 result = EFFECT_SESSION; 1908 } 1909 1910 for (size_t i = 0; i < mTracks.size(); ++i) { 1911 sp<Track> track = mTracks[i]; 1912 if (sessionId == track->sessionId() && 1913 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1914 result |= TRACK_SESSION; 1915 break; 1916 } 1917 } 1918 1919 return result; 1920} 1921 1922uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1923{ 1924 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1925 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1926 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1927 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1928 } 1929 for (size_t i = 0; i < mTracks.size(); i++) { 1930 sp<Track> track = mTracks[i]; 1931 if (sessionId == track->sessionId() && 1932 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1933 return AudioSystem::getStrategyForStream(track->streamType()); 1934 } 1935 } 1936 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1937} 1938 1939 1940AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1941{ 1942 Mutex::Autolock _l(mLock); 1943 return mOutput; 1944} 1945 1946AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1947{ 1948 Mutex::Autolock _l(mLock); 1949 AudioStreamOut *output = mOutput; 1950 mOutput = NULL; 1951 return output; 1952} 1953 1954// this method must always be called either with ThreadBase mLock held or inside the thread loop 1955audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1956{ 1957 if (mOutput == NULL) { 1958 return NULL; 1959 } 1960 return &mOutput->stream->common; 1961} 1962 1963uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1964{ 1965 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1966 // decoding and transfer time. So sleeping for half of the latency would likely cause 1967 // underruns 1968 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1969 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1970 } else { 1971 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1972 } 1973} 1974 1975status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1976{ 1977 if (!isValidSyncEvent(event)) { 1978 return BAD_VALUE; 1979 } 1980 1981 Mutex::Autolock _l(mLock); 1982 1983 for (size_t i = 0; i < mTracks.size(); ++i) { 1984 sp<Track> track = mTracks[i]; 1985 if (event->triggerSession() == track->sessionId()) { 1986 track->setSyncEvent(event); 1987 return NO_ERROR; 1988 } 1989 } 1990 1991 return NAME_NOT_FOUND; 1992} 1993 1994bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 1995{ 1996 switch (event->type()) { 1997 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 1998 return true; 1999 default: 2000 break; 2001 } 2002 return false; 2003} 2004 2005// ---------------------------------------------------------------------------- 2006 2007AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2008 audio_io_handle_t id, uint32_t device, type_t type) 2009 : PlaybackThread(audioFlinger, output, id, device, type) 2010{ 2011 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2012 // FIXME - Current mixer implementation only supports stereo output 2013 if (mChannelCount == 1) { 2014 ALOGE("Invalid audio hardware channel count"); 2015 } 2016} 2017 2018AudioFlinger::MixerThread::~MixerThread() 2019{ 2020 delete mAudioMixer; 2021} 2022 2023class CpuStats { 2024public: 2025 CpuStats(); 2026 void sample(const String8 &title); 2027#ifdef DEBUG_CPU_USAGE 2028private: 2029 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2030 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2031 2032 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2033 2034 int mCpuNum; // thread's current CPU number 2035 int mCpukHz; // frequency of thread's current CPU in kHz 2036#endif 2037}; 2038 2039CpuStats::CpuStats() 2040#ifdef DEBUG_CPU_USAGE 2041 : mCpuNum(-1), mCpukHz(-1) 2042#endif 2043{ 2044} 2045 2046void CpuStats::sample(const String8 &title) { 2047#ifdef DEBUG_CPU_USAGE 2048 // get current thread's delta CPU time in wall clock ns 2049 double wcNs; 2050 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2051 2052 // record sample for wall clock statistics 2053 if (valid) { 2054 mWcStats.sample(wcNs); 2055 } 2056 2057 // get the current CPU number 2058 int cpuNum = sched_getcpu(); 2059 2060 // get the current CPU frequency in kHz 2061 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2062 2063 // check if either CPU number or frequency changed 2064 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2065 mCpuNum = cpuNum; 2066 mCpukHz = cpukHz; 2067 // ignore sample for purposes of cycles 2068 valid = false; 2069 } 2070 2071 // if no change in CPU number or frequency, then record sample for cycle statistics 2072 if (valid && mCpukHz > 0) { 2073 double cycles = wcNs * cpukHz * 0.000001; 2074 mHzStats.sample(cycles); 2075 } 2076 2077 unsigned n = mWcStats.n(); 2078 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2079 if ((n & 127) == 1) { 2080 long long elapsed = mCpuUsage.elapsed(); 2081 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2082 double perLoop = elapsed / (double) n; 2083 double perLoop100 = perLoop * 0.01; 2084 double perLoop1k = perLoop * 0.001; 2085 double mean = mWcStats.mean(); 2086 double stddev = mWcStats.stddev(); 2087 double minimum = mWcStats.minimum(); 2088 double maximum = mWcStats.maximum(); 2089 double meanCycles = mHzStats.mean(); 2090 double stddevCycles = mHzStats.stddev(); 2091 double minCycles = mHzStats.minimum(); 2092 double maxCycles = mHzStats.maximum(); 2093 mCpuUsage.resetElapsed(); 2094 mWcStats.reset(); 2095 mHzStats.reset(); 2096 ALOGD("CPU usage for %s over past %.1f secs\n" 2097 " (%u mixer loops at %.1f mean ms per loop):\n" 2098 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2099 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2100 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2101 title.string(), 2102 elapsed * .000000001, n, perLoop * .000001, 2103 mean * .001, 2104 stddev * .001, 2105 minimum * .001, 2106 maximum * .001, 2107 mean / perLoop100, 2108 stddev / perLoop100, 2109 minimum / perLoop100, 2110 maximum / perLoop100, 2111 meanCycles / perLoop1k, 2112 stddevCycles / perLoop1k, 2113 minCycles / perLoop1k, 2114 maxCycles / perLoop1k); 2115 2116 } 2117 } 2118#endif 2119}; 2120 2121void AudioFlinger::PlaybackThread::checkSilentMode_l() 2122{ 2123 if (!mMasterMute) { 2124 char value[PROPERTY_VALUE_MAX]; 2125 if (property_get("ro.audio.silent", value, "0") > 0) { 2126 char *endptr; 2127 unsigned long ul = strtoul(value, &endptr, 0); 2128 if (*endptr == '\0' && ul != 0) { 2129 ALOGD("Silence is golden"); 2130 // The setprop command will not allow a property to be changed after 2131 // the first time it is set, so we don't have to worry about un-muting. 2132 setMasterMute_l(true); 2133 } 2134 } 2135 } 2136} 2137 2138bool AudioFlinger::PlaybackThread::threadLoop() 2139{ 2140 Vector< sp<Track> > tracksToRemove; 2141 2142 standbyTime = systemTime(); 2143 2144 // MIXER 2145 nsecs_t lastWarning = 0; 2146if (mType == MIXER) { 2147 longStandbyExit = false; 2148} 2149 2150 // DUPLICATING 2151 // FIXME could this be made local to while loop? 2152 writeFrames = 0; 2153 2154 cacheParameters_l(); 2155 sleepTime = idleSleepTime; 2156 2157if (mType == MIXER) { 2158 sleepTimeShift = 0; 2159} 2160 2161 CpuStats cpuStats; 2162 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2163 2164 acquireWakeLock(); 2165 2166 while (!exitPending()) 2167 { 2168 cpuStats.sample(myName); 2169 2170 Vector< sp<EffectChain> > effectChains; 2171 2172 processConfigEvents(); 2173 2174 { // scope for mLock 2175 2176 Mutex::Autolock _l(mLock); 2177 2178 if (checkForNewParameters_l()) { 2179 cacheParameters_l(); 2180 } 2181 2182 saveOutputTracks(); 2183 2184 // put audio hardware into standby after short delay 2185 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2186 mSuspended > 0)) { 2187 if (!mStandby) { 2188 2189 threadLoop_standby(); 2190 2191 mStandby = true; 2192 mBytesWritten = 0; 2193 } 2194 2195 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2196 // we're about to wait, flush the binder command buffer 2197 IPCThreadState::self()->flushCommands(); 2198 2199 clearOutputTracks(); 2200 2201 if (exitPending()) break; 2202 2203 releaseWakeLock_l(); 2204 // wait until we have something to do... 2205 ALOGV("%s going to sleep", myName.string()); 2206 mWaitWorkCV.wait(mLock); 2207 ALOGV("%s waking up", myName.string()); 2208 acquireWakeLock_l(); 2209 2210 mPrevMixerStatus = MIXER_IDLE; 2211 2212 checkSilentMode_l(); 2213 2214 standbyTime = systemTime() + standbyDelay; 2215 sleepTime = idleSleepTime; 2216 if (mType == MIXER) { 2217 sleepTimeShift = 0; 2218 } 2219 2220 continue; 2221 } 2222 } 2223 2224 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2225 // Shift in the new status; this could be a queue if it's 2226 // useful to filter the mixer status over several cycles. 2227 mPrevMixerStatus = mMixerStatus; 2228 mMixerStatus = newMixerStatus; 2229 2230 // prevent any changes in effect chain list and in each effect chain 2231 // during mixing and effect process as the audio buffers could be deleted 2232 // or modified if an effect is created or deleted 2233 lockEffectChains_l(effectChains); 2234 } 2235 2236 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2237 threadLoop_mix(); 2238 } else { 2239 threadLoop_sleepTime(); 2240 } 2241 2242 if (mSuspended > 0) { 2243 sleepTime = suspendSleepTimeUs(); 2244 } 2245 2246 // only process effects if we're going to write 2247 if (sleepTime == 0) { 2248 for (size_t i = 0; i < effectChains.size(); i ++) { 2249 effectChains[i]->process_l(); 2250 } 2251 } 2252 2253 // enable changes in effect chain 2254 unlockEffectChains(effectChains); 2255 2256 // sleepTime == 0 means we must write to audio hardware 2257 if (sleepTime == 0) { 2258 2259 threadLoop_write(); 2260 2261if (mType == MIXER) { 2262 // write blocked detection 2263 nsecs_t now = systemTime(); 2264 nsecs_t delta = now - mLastWriteTime; 2265 if (!mStandby && delta > maxPeriod) { 2266 mNumDelayedWrites++; 2267 if ((now - lastWarning) > kWarningThrottleNs) { 2268 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2269 ns2ms(delta), mNumDelayedWrites, this); 2270 lastWarning = now; 2271 } 2272 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2273 // a different threshold. Or completely removed for what it is worth anyway... 2274 if (mStandby) { 2275 longStandbyExit = true; 2276 } 2277 } 2278} 2279 2280 mStandby = false; 2281 } else { 2282 usleep(sleepTime); 2283 } 2284 2285 // finally let go of removed track(s), without the lock held 2286 // since we can't guarantee the destructors won't acquire that 2287 // same lock. 2288 tracksToRemove.clear(); 2289 2290 // FIXME I don't understand the need for this here; 2291 // it was in the original code but maybe the 2292 // assignment in saveOutputTracks() makes this unnecessary? 2293 clearOutputTracks(); 2294 2295 // Effect chains will be actually deleted here if they were removed from 2296 // mEffectChains list during mixing or effects processing 2297 effectChains.clear(); 2298 2299 // FIXME Note that the above .clear() is no longer necessary since effectChains 2300 // is now local to this block, but will keep it for now (at least until merge done). 2301 } 2302 2303if (mType == MIXER || mType == DIRECT) { 2304 // put output stream into standby mode 2305 if (!mStandby) { 2306 mOutput->stream->common.standby(&mOutput->stream->common); 2307 } 2308} 2309if (mType == DUPLICATING) { 2310 // for DuplicatingThread, standby mode is handled by the outputTracks 2311} 2312 2313 releaseWakeLock(); 2314 2315 ALOGV("Thread %p type %d exiting", this, mType); 2316 return false; 2317} 2318 2319// shared by MIXER and DIRECT, overridden by DUPLICATING 2320void AudioFlinger::PlaybackThread::threadLoop_write() 2321{ 2322 // FIXME rewrite to reduce number of system calls 2323 mLastWriteTime = systemTime(); 2324 mInWrite = true; 2325 mBytesWritten += mixBufferSize; 2326 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2327 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2328 mNumWrites++; 2329 mInWrite = false; 2330} 2331 2332// shared by MIXER and DIRECT, overridden by DUPLICATING 2333void AudioFlinger::PlaybackThread::threadLoop_standby() 2334{ 2335 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2336 mOutput->stream->common.standby(&mOutput->stream->common); 2337} 2338 2339void AudioFlinger::MixerThread::threadLoop_mix() 2340{ 2341 // obtain the presentation timestamp of the next output buffer 2342 int64_t pts; 2343 status_t status = INVALID_OPERATION; 2344 2345 if (NULL != mOutput->stream->get_next_write_timestamp) { 2346 status = mOutput->stream->get_next_write_timestamp( 2347 mOutput->stream, &pts); 2348 } 2349 2350 if (status != NO_ERROR) { 2351 pts = AudioBufferProvider::kInvalidPTS; 2352 } 2353 2354 // mix buffers... 2355 mAudioMixer->process(pts); 2356 // increase sleep time progressively when application underrun condition clears. 2357 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2358 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2359 // such that we would underrun the audio HAL. 2360 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2361 sleepTimeShift--; 2362 } 2363 sleepTime = 0; 2364 standbyTime = systemTime() + standbyDelay; 2365 //TODO: delay standby when effects have a tail 2366} 2367 2368void AudioFlinger::MixerThread::threadLoop_sleepTime() 2369{ 2370 // If no tracks are ready, sleep once for the duration of an output 2371 // buffer size, then write 0s to the output 2372 if (sleepTime == 0) { 2373 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2374 sleepTime = activeSleepTime >> sleepTimeShift; 2375 if (sleepTime < kMinThreadSleepTimeUs) { 2376 sleepTime = kMinThreadSleepTimeUs; 2377 } 2378 // reduce sleep time in case of consecutive application underruns to avoid 2379 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2380 // duration we would end up writing less data than needed by the audio HAL if 2381 // the condition persists. 2382 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2383 sleepTimeShift++; 2384 } 2385 } else { 2386 sleepTime = idleSleepTime; 2387 } 2388 } else if (mBytesWritten != 0 || 2389 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2390 memset (mMixBuffer, 0, mixBufferSize); 2391 sleepTime = 0; 2392 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2393 } 2394 // TODO add standby time extension fct of effect tail 2395} 2396 2397// prepareTracks_l() must be called with ThreadBase::mLock held 2398AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2399 Vector< sp<Track> > *tracksToRemove) 2400{ 2401 2402 mixer_state mixerStatus = MIXER_IDLE; 2403 // find out which tracks need to be processed 2404 size_t count = mActiveTracks.size(); 2405 size_t mixedTracks = 0; 2406 size_t tracksWithEffect = 0; 2407 2408 float masterVolume = mMasterVolume; 2409 bool masterMute = mMasterMute; 2410 2411 if (masterMute) { 2412 masterVolume = 0; 2413 } 2414 // Delegate master volume control to effect in output mix effect chain if needed 2415 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2416 if (chain != 0) { 2417 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2418 chain->setVolume_l(&v, &v); 2419 masterVolume = (float)((v + (1 << 23)) >> 24); 2420 chain.clear(); 2421 } 2422 2423 for (size_t i=0 ; i<count ; i++) { 2424 sp<Track> t = mActiveTracks[i].promote(); 2425 if (t == 0) continue; 2426 2427 // this const just means the local variable doesn't change 2428 Track* const track = t.get(); 2429 audio_track_cblk_t* cblk = track->cblk(); 2430 2431 // The first time a track is added we wait 2432 // for all its buffers to be filled before processing it 2433 int name = track->name(); 2434 // make sure that we have enough frames to mix one full buffer. 2435 // enforce this condition only once to enable draining the buffer in case the client 2436 // app does not call stop() and relies on underrun to stop: 2437 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2438 // during last round 2439 uint32_t minFrames = 1; 2440 if (!track->isStopped() && !track->isPausing() && 2441 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2442 if (t->sampleRate() == (int)mSampleRate) { 2443 minFrames = mFrameCount; 2444 } else { 2445 // +1 for rounding and +1 for additional sample needed for interpolation 2446 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2447 // add frames already consumed but not yet released by the resampler 2448 // because cblk->framesReady() will include these frames 2449 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2450 // the minimum track buffer size is normally twice the number of frames necessary 2451 // to fill one buffer and the resampler should not leave more than one buffer worth 2452 // of unreleased frames after each pass, but just in case... 2453 ALOG_ASSERT(minFrames <= cblk->frameCount); 2454 } 2455 } 2456 if ((track->framesReady() >= minFrames) && track->isReady() && 2457 !track->isPaused() && !track->isTerminated()) 2458 { 2459 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2460 2461 mixedTracks++; 2462 2463 // track->mainBuffer() != mMixBuffer means there is an effect chain 2464 // connected to the track 2465 chain.clear(); 2466 if (track->mainBuffer() != mMixBuffer) { 2467 chain = getEffectChain_l(track->sessionId()); 2468 // Delegate volume control to effect in track effect chain if needed 2469 if (chain != 0) { 2470 tracksWithEffect++; 2471 } else { 2472 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2473 name, track->sessionId()); 2474 } 2475 } 2476 2477 2478 int param = AudioMixer::VOLUME; 2479 if (track->mFillingUpStatus == Track::FS_FILLED) { 2480 // no ramp for the first volume setting 2481 track->mFillingUpStatus = Track::FS_ACTIVE; 2482 if (track->mState == TrackBase::RESUMING) { 2483 track->mState = TrackBase::ACTIVE; 2484 param = AudioMixer::RAMP_VOLUME; 2485 } 2486 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2487 } else if (cblk->server != 0) { 2488 // If the track is stopped before the first frame was mixed, 2489 // do not apply ramp 2490 param = AudioMixer::RAMP_VOLUME; 2491 } 2492 2493 // compute volume for this track 2494 uint32_t vl, vr, va; 2495 if (track->isMuted() || track->isPausing() || 2496 mStreamTypes[track->streamType()].mute) { 2497 vl = vr = va = 0; 2498 if (track->isPausing()) { 2499 track->setPaused(); 2500 } 2501 } else { 2502 2503 // read original volumes with volume control 2504 float typeVolume = mStreamTypes[track->streamType()].volume; 2505 float v = masterVolume * typeVolume; 2506 uint32_t vlr = cblk->getVolumeLR(); 2507 vl = vlr & 0xFFFF; 2508 vr = vlr >> 16; 2509 // track volumes come from shared memory, so can't be trusted and must be clamped 2510 if (vl > MAX_GAIN_INT) { 2511 ALOGV("Track left volume out of range: %04X", vl); 2512 vl = MAX_GAIN_INT; 2513 } 2514 if (vr > MAX_GAIN_INT) { 2515 ALOGV("Track right volume out of range: %04X", vr); 2516 vr = MAX_GAIN_INT; 2517 } 2518 // now apply the master volume and stream type volume 2519 vl = (uint32_t)(v * vl) << 12; 2520 vr = (uint32_t)(v * vr) << 12; 2521 // assuming master volume and stream type volume each go up to 1.0, 2522 // vl and vr are now in 8.24 format 2523 2524 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2525 // send level comes from shared memory and so may be corrupt 2526 if (sendLevel > MAX_GAIN_INT) { 2527 ALOGV("Track send level out of range: %04X", sendLevel); 2528 sendLevel = MAX_GAIN_INT; 2529 } 2530 va = (uint32_t)(v * sendLevel); 2531 } 2532 // Delegate volume control to effect in track effect chain if needed 2533 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2534 // Do not ramp volume if volume is controlled by effect 2535 param = AudioMixer::VOLUME; 2536 track->mHasVolumeController = true; 2537 } else { 2538 // force no volume ramp when volume controller was just disabled or removed 2539 // from effect chain to avoid volume spike 2540 if (track->mHasVolumeController) { 2541 param = AudioMixer::VOLUME; 2542 } 2543 track->mHasVolumeController = false; 2544 } 2545 2546 // Convert volumes from 8.24 to 4.12 format 2547 // This additional clamping is needed in case chain->setVolume_l() overshot 2548 vl = (vl + (1 << 11)) >> 12; 2549 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2550 vr = (vr + (1 << 11)) >> 12; 2551 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2552 2553 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2554 2555 // XXX: these things DON'T need to be done each time 2556 mAudioMixer->setBufferProvider(name, track); 2557 mAudioMixer->enable(name); 2558 2559 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2560 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2561 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2562 mAudioMixer->setParameter( 2563 name, 2564 AudioMixer::TRACK, 2565 AudioMixer::FORMAT, (void *)track->format()); 2566 mAudioMixer->setParameter( 2567 name, 2568 AudioMixer::TRACK, 2569 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2570 mAudioMixer->setParameter( 2571 name, 2572 AudioMixer::RESAMPLE, 2573 AudioMixer::SAMPLE_RATE, 2574 (void *)(cblk->sampleRate)); 2575 mAudioMixer->setParameter( 2576 name, 2577 AudioMixer::TRACK, 2578 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2579 mAudioMixer->setParameter( 2580 name, 2581 AudioMixer::TRACK, 2582 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2583 2584 // reset retry count 2585 track->mRetryCount = kMaxTrackRetries; 2586 2587 // If one track is ready, set the mixer ready if: 2588 // - the mixer was not ready during previous round OR 2589 // - no other track is not ready 2590 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2591 mixerStatus != MIXER_TRACKS_ENABLED) { 2592 mixerStatus = MIXER_TRACKS_READY; 2593 } 2594 } else { 2595 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2596 if (track->isStopped()) { 2597 track->reset(); 2598 } 2599 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2600 // We have consumed all the buffers of this track. 2601 // Remove it from the list of active tracks. 2602 // TODO: use actual buffer filling status instead of latency when available from 2603 // audio HAL 2604 size_t audioHALFrames = 2605 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2606 size_t framesWritten = 2607 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2608 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2609 tracksToRemove->add(track); 2610 } 2611 } else { 2612 // No buffers for this track. Give it a few chances to 2613 // fill a buffer, then remove it from active list. 2614 if (--(track->mRetryCount) <= 0) { 2615 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2616 tracksToRemove->add(track); 2617 // indicate to client process that the track was disabled because of underrun 2618 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2619 // If one track is not ready, mark the mixer also not ready if: 2620 // - the mixer was ready during previous round OR 2621 // - no other track is ready 2622 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2623 mixerStatus != MIXER_TRACKS_READY) { 2624 mixerStatus = MIXER_TRACKS_ENABLED; 2625 } 2626 } 2627 mAudioMixer->disable(name); 2628 } 2629 } 2630 2631 // remove all the tracks that need to be... 2632 count = tracksToRemove->size(); 2633 if (CC_UNLIKELY(count)) { 2634 for (size_t i=0 ; i<count ; i++) { 2635 const sp<Track>& track = tracksToRemove->itemAt(i); 2636 mActiveTracks.remove(track); 2637 if (track->mainBuffer() != mMixBuffer) { 2638 chain = getEffectChain_l(track->sessionId()); 2639 if (chain != 0) { 2640 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2641 chain->decActiveTrackCnt(); 2642 } 2643 } 2644 if (track->isTerminated()) { 2645 removeTrack_l(track); 2646 } 2647 } 2648 } 2649 2650 // mix buffer must be cleared if all tracks are connected to an 2651 // effect chain as in this case the mixer will not write to 2652 // mix buffer and track effects will accumulate into it 2653 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2654 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2655 } 2656 2657 return mixerStatus; 2658} 2659 2660/* 2661The derived values that are cached: 2662 - mixBufferSize from frame count * frame size 2663 - activeSleepTime from activeSleepTimeUs() 2664 - idleSleepTime from idleSleepTimeUs() 2665 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2666 - maxPeriod from frame count and sample rate (MIXER only) 2667 2668The parameters that affect these derived values are: 2669 - frame count 2670 - frame size 2671 - sample rate 2672 - device type: A2DP or not 2673 - device latency 2674 - format: PCM or not 2675 - active sleep time 2676 - idle sleep time 2677*/ 2678 2679void AudioFlinger::PlaybackThread::cacheParameters_l() 2680{ 2681 mixBufferSize = mFrameCount * mFrameSize; 2682 activeSleepTime = activeSleepTimeUs(); 2683 idleSleepTime = idleSleepTimeUs(); 2684} 2685 2686void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2687{ 2688 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2689 this, streamType, mTracks.size()); 2690 Mutex::Autolock _l(mLock); 2691 2692 size_t size = mTracks.size(); 2693 for (size_t i = 0; i < size; i++) { 2694 sp<Track> t = mTracks[i]; 2695 if (t->streamType() == streamType) { 2696 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2697 t->mCblk->cv.signal(); 2698 } 2699 } 2700} 2701 2702// getTrackName_l() must be called with ThreadBase::mLock held 2703int AudioFlinger::MixerThread::getTrackName_l() 2704{ 2705 return mAudioMixer->getTrackName(); 2706} 2707 2708// deleteTrackName_l() must be called with ThreadBase::mLock held 2709void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2710{ 2711 ALOGV("remove track (%d) and delete from mixer", name); 2712 mAudioMixer->deleteTrackName(name); 2713} 2714 2715// checkForNewParameters_l() must be called with ThreadBase::mLock held 2716bool AudioFlinger::MixerThread::checkForNewParameters_l() 2717{ 2718 bool reconfig = false; 2719 2720 while (!mNewParameters.isEmpty()) { 2721 status_t status = NO_ERROR; 2722 String8 keyValuePair = mNewParameters[0]; 2723 AudioParameter param = AudioParameter(keyValuePair); 2724 int value; 2725 2726 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2727 reconfig = true; 2728 } 2729 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2730 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2731 status = BAD_VALUE; 2732 } else { 2733 reconfig = true; 2734 } 2735 } 2736 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2737 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2738 status = BAD_VALUE; 2739 } else { 2740 reconfig = true; 2741 } 2742 } 2743 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2744 // do not accept frame count changes if tracks are open as the track buffer 2745 // size depends on frame count and correct behavior would not be guaranteed 2746 // if frame count is changed after track creation 2747 if (!mTracks.isEmpty()) { 2748 status = INVALID_OPERATION; 2749 } else { 2750 reconfig = true; 2751 } 2752 } 2753 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2754#ifdef ADD_BATTERY_DATA 2755 // when changing the audio output device, call addBatteryData to notify 2756 // the change 2757 if ((int)mDevice != value) { 2758 uint32_t params = 0; 2759 // check whether speaker is on 2760 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2761 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2762 } 2763 2764 int deviceWithoutSpeaker 2765 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2766 // check if any other device (except speaker) is on 2767 if (value & deviceWithoutSpeaker ) { 2768 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2769 } 2770 2771 if (params != 0) { 2772 addBatteryData(params); 2773 } 2774 } 2775#endif 2776 2777 // forward device change to effects that have requested to be 2778 // aware of attached audio device. 2779 mDevice = (uint32_t)value; 2780 for (size_t i = 0; i < mEffectChains.size(); i++) { 2781 mEffectChains[i]->setDevice_l(mDevice); 2782 } 2783 } 2784 2785 if (status == NO_ERROR) { 2786 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2787 keyValuePair.string()); 2788 if (!mStandby && status == INVALID_OPERATION) { 2789 mOutput->stream->common.standby(&mOutput->stream->common); 2790 mStandby = true; 2791 mBytesWritten = 0; 2792 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2793 keyValuePair.string()); 2794 } 2795 if (status == NO_ERROR && reconfig) { 2796 delete mAudioMixer; 2797 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2798 mAudioMixer = NULL; 2799 readOutputParameters(); 2800 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2801 for (size_t i = 0; i < mTracks.size() ; i++) { 2802 int name = getTrackName_l(); 2803 if (name < 0) break; 2804 mTracks[i]->mName = name; 2805 // limit track sample rate to 2 x new output sample rate 2806 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2807 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2808 } 2809 } 2810 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2811 } 2812 } 2813 2814 mNewParameters.removeAt(0); 2815 2816 mParamStatus = status; 2817 mParamCond.signal(); 2818 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2819 // already timed out waiting for the status and will never signal the condition. 2820 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2821 } 2822 return reconfig; 2823} 2824 2825status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2826{ 2827 const size_t SIZE = 256; 2828 char buffer[SIZE]; 2829 String8 result; 2830 2831 PlaybackThread::dumpInternals(fd, args); 2832 2833 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2834 result.append(buffer); 2835 write(fd, result.string(), result.size()); 2836 return NO_ERROR; 2837} 2838 2839uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 2840{ 2841 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2842} 2843 2844uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 2845{ 2846 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2847} 2848 2849void AudioFlinger::MixerThread::cacheParameters_l() 2850{ 2851 PlaybackThread::cacheParameters_l(); 2852 2853 // FIXME: Relaxed timing because of a certain device that can't meet latency 2854 // Should be reduced to 2x after the vendor fixes the driver issue 2855 // increase threshold again due to low power audio mode. The way this warning 2856 // threshold is calculated and its usefulness should be reconsidered anyway. 2857 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2858} 2859 2860// ---------------------------------------------------------------------------- 2861AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2862 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2863 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2864 // mLeftVolFloat, mRightVolFloat 2865 // mLeftVolShort, mRightVolShort 2866{ 2867} 2868 2869AudioFlinger::DirectOutputThread::~DirectOutputThread() 2870{ 2871} 2872 2873AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2874 Vector< sp<Track> > *tracksToRemove 2875) 2876{ 2877 sp<Track> trackToRemove; 2878 2879 mixer_state mixerStatus = MIXER_IDLE; 2880 2881 // find out which tracks need to be processed 2882 if (mActiveTracks.size() != 0) { 2883 sp<Track> t = mActiveTracks[0].promote(); 2884 // The track died recently 2885 if (t == 0) return MIXER_IDLE; 2886 2887 Track* const track = t.get(); 2888 audio_track_cblk_t* cblk = track->cblk(); 2889 2890 // The first time a track is added we wait 2891 // for all its buffers to be filled before processing it 2892 if (cblk->framesReady() && track->isReady() && 2893 !track->isPaused() && !track->isTerminated()) 2894 { 2895 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2896 2897 if (track->mFillingUpStatus == Track::FS_FILLED) { 2898 track->mFillingUpStatus = Track::FS_ACTIVE; 2899 mLeftVolFloat = mRightVolFloat = 0; 2900 mLeftVolShort = mRightVolShort = 0; 2901 if (track->mState == TrackBase::RESUMING) { 2902 track->mState = TrackBase::ACTIVE; 2903 rampVolume = true; 2904 } 2905 } else if (cblk->server != 0) { 2906 // If the track is stopped before the first frame was mixed, 2907 // do not apply ramp 2908 rampVolume = true; 2909 } 2910 // compute volume for this track 2911 float left, right; 2912 if (track->isMuted() || mMasterMute || track->isPausing() || 2913 mStreamTypes[track->streamType()].mute) { 2914 left = right = 0; 2915 if (track->isPausing()) { 2916 track->setPaused(); 2917 } 2918 } else { 2919 float typeVolume = mStreamTypes[track->streamType()].volume; 2920 float v = mMasterVolume * typeVolume; 2921 uint32_t vlr = cblk->getVolumeLR(); 2922 float v_clamped = v * (vlr & 0xFFFF); 2923 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2924 left = v_clamped/MAX_GAIN; 2925 v_clamped = v * (vlr >> 16); 2926 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2927 right = v_clamped/MAX_GAIN; 2928 } 2929 2930 if (left != mLeftVolFloat || right != mRightVolFloat) { 2931 mLeftVolFloat = left; 2932 mRightVolFloat = right; 2933 2934 // If audio HAL implements volume control, 2935 // force software volume to nominal value 2936 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2937 left = 1.0f; 2938 right = 1.0f; 2939 } 2940 2941 // Convert volumes from float to 8.24 2942 uint32_t vl = (uint32_t)(left * (1 << 24)); 2943 uint32_t vr = (uint32_t)(right * (1 << 24)); 2944 2945 // Delegate volume control to effect in track effect chain if needed 2946 // only one effect chain can be present on DirectOutputThread, so if 2947 // there is one, the track is connected to it 2948 if (!mEffectChains.isEmpty()) { 2949 // Do not ramp volume if volume is controlled by effect 2950 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2951 rampVolume = false; 2952 } 2953 } 2954 2955 // Convert volumes from 8.24 to 4.12 format 2956 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2957 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2958 leftVol = (uint16_t)v_clamped; 2959 v_clamped = (vr + (1 << 11)) >> 12; 2960 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2961 rightVol = (uint16_t)v_clamped; 2962 } else { 2963 leftVol = mLeftVolShort; 2964 rightVol = mRightVolShort; 2965 rampVolume = false; 2966 } 2967 2968 // reset retry count 2969 track->mRetryCount = kMaxTrackRetriesDirect; 2970 mActiveTrack = t; 2971 mixerStatus = MIXER_TRACKS_READY; 2972 } else { 2973 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2974 if (track->isStopped()) { 2975 track->reset(); 2976 } 2977 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2978 // We have consumed all the buffers of this track. 2979 // Remove it from the list of active tracks. 2980 // TODO: implement behavior for compressed audio 2981 size_t audioHALFrames = 2982 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2983 size_t framesWritten = 2984 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2985 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2986 trackToRemove = track; 2987 } 2988 } else { 2989 // No buffers for this track. Give it a few chances to 2990 // fill a buffer, then remove it from active list. 2991 if (--(track->mRetryCount) <= 0) { 2992 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2993 trackToRemove = track; 2994 } else { 2995 mixerStatus = MIXER_TRACKS_ENABLED; 2996 } 2997 } 2998 } 2999 } 3000 3001 // FIXME merge this with similar code for removing multiple tracks 3002 // remove all the tracks that need to be... 3003 if (CC_UNLIKELY(trackToRemove != 0)) { 3004 tracksToRemove->add(trackToRemove); 3005 mActiveTracks.remove(trackToRemove); 3006 if (!mEffectChains.isEmpty()) { 3007 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3008 trackToRemove->sessionId()); 3009 mEffectChains[0]->decActiveTrackCnt(); 3010 } 3011 if (trackToRemove->isTerminated()) { 3012 removeTrack_l(trackToRemove); 3013 } 3014 } 3015 3016 return mixerStatus; 3017} 3018 3019void AudioFlinger::DirectOutputThread::threadLoop_mix() 3020{ 3021 AudioBufferProvider::Buffer buffer; 3022 size_t frameCount = mFrameCount; 3023 int8_t *curBuf = (int8_t *)mMixBuffer; 3024 // output audio to hardware 3025 while (frameCount) { 3026 buffer.frameCount = frameCount; 3027 mActiveTrack->getNextBuffer(&buffer); 3028 if (CC_UNLIKELY(buffer.raw == NULL)) { 3029 memset(curBuf, 0, frameCount * mFrameSize); 3030 break; 3031 } 3032 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3033 frameCount -= buffer.frameCount; 3034 curBuf += buffer.frameCount * mFrameSize; 3035 mActiveTrack->releaseBuffer(&buffer); 3036 } 3037 sleepTime = 0; 3038 standbyTime = systemTime() + standbyDelay; 3039 mActiveTrack.clear(); 3040 3041 // apply volume 3042 3043 // Do not apply volume on compressed audio 3044 if (!audio_is_linear_pcm(mFormat)) { 3045 return; 3046 } 3047 3048 // convert to signed 16 bit before volume calculation 3049 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3050 size_t count = mFrameCount * mChannelCount; 3051 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3052 int16_t *dst = mMixBuffer + count-1; 3053 while (count--) { 3054 *dst-- = (int16_t)(*src--^0x80) << 8; 3055 } 3056 } 3057 3058 frameCount = mFrameCount; 3059 int16_t *out = mMixBuffer; 3060 if (rampVolume) { 3061 if (mChannelCount == 1) { 3062 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3063 int32_t vlInc = d / (int32_t)frameCount; 3064 int32_t vl = ((int32_t)mLeftVolShort << 16); 3065 do { 3066 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3067 out++; 3068 vl += vlInc; 3069 } while (--frameCount); 3070 3071 } else { 3072 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3073 int32_t vlInc = d / (int32_t)frameCount; 3074 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3075 int32_t vrInc = d / (int32_t)frameCount; 3076 int32_t vl = ((int32_t)mLeftVolShort << 16); 3077 int32_t vr = ((int32_t)mRightVolShort << 16); 3078 do { 3079 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3080 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3081 out += 2; 3082 vl += vlInc; 3083 vr += vrInc; 3084 } while (--frameCount); 3085 } 3086 } else { 3087 if (mChannelCount == 1) { 3088 do { 3089 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3090 out++; 3091 } while (--frameCount); 3092 } else { 3093 do { 3094 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3095 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3096 out += 2; 3097 } while (--frameCount); 3098 } 3099 } 3100 3101 // convert back to unsigned 8 bit after volume calculation 3102 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3103 size_t count = mFrameCount * mChannelCount; 3104 int16_t *src = mMixBuffer; 3105 uint8_t *dst = (uint8_t *)mMixBuffer; 3106 while (count--) { 3107 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3108 } 3109 } 3110 3111 mLeftVolShort = leftVol; 3112 mRightVolShort = rightVol; 3113} 3114 3115void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3116{ 3117 if (sleepTime == 0) { 3118 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3119 sleepTime = activeSleepTime; 3120 } else { 3121 sleepTime = idleSleepTime; 3122 } 3123 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3124 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3125 sleepTime = 0; 3126 } 3127} 3128 3129// getTrackName_l() must be called with ThreadBase::mLock held 3130int AudioFlinger::DirectOutputThread::getTrackName_l() 3131{ 3132 return 0; 3133} 3134 3135// deleteTrackName_l() must be called with ThreadBase::mLock held 3136void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3137{ 3138} 3139 3140// checkForNewParameters_l() must be called with ThreadBase::mLock held 3141bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3142{ 3143 bool reconfig = false; 3144 3145 while (!mNewParameters.isEmpty()) { 3146 status_t status = NO_ERROR; 3147 String8 keyValuePair = mNewParameters[0]; 3148 AudioParameter param = AudioParameter(keyValuePair); 3149 int value; 3150 3151 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3152 // do not accept frame count changes if tracks are open as the track buffer 3153 // size depends on frame count and correct behavior would not be garantied 3154 // if frame count is changed after track creation 3155 if (!mTracks.isEmpty()) { 3156 status = INVALID_OPERATION; 3157 } else { 3158 reconfig = true; 3159 } 3160 } 3161 if (status == NO_ERROR) { 3162 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3163 keyValuePair.string()); 3164 if (!mStandby && status == INVALID_OPERATION) { 3165 mOutput->stream->common.standby(&mOutput->stream->common); 3166 mStandby = true; 3167 mBytesWritten = 0; 3168 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3169 keyValuePair.string()); 3170 } 3171 if (status == NO_ERROR && reconfig) { 3172 readOutputParameters(); 3173 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3174 } 3175 } 3176 3177 mNewParameters.removeAt(0); 3178 3179 mParamStatus = status; 3180 mParamCond.signal(); 3181 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3182 // already timed out waiting for the status and will never signal the condition. 3183 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3184 } 3185 return reconfig; 3186} 3187 3188uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3189{ 3190 uint32_t time; 3191 if (audio_is_linear_pcm(mFormat)) { 3192 time = PlaybackThread::activeSleepTimeUs(); 3193 } else { 3194 time = 10000; 3195 } 3196 return time; 3197} 3198 3199uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3200{ 3201 uint32_t time; 3202 if (audio_is_linear_pcm(mFormat)) { 3203 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3204 } else { 3205 time = 10000; 3206 } 3207 return time; 3208} 3209 3210uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3211{ 3212 uint32_t time; 3213 if (audio_is_linear_pcm(mFormat)) { 3214 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3215 } else { 3216 time = 10000; 3217 } 3218 return time; 3219} 3220 3221void AudioFlinger::DirectOutputThread::cacheParameters_l() 3222{ 3223 PlaybackThread::cacheParameters_l(); 3224 3225 // use shorter standby delay as on normal output to release 3226 // hardware resources as soon as possible 3227 standbyDelay = microseconds(activeSleepTime*2); 3228} 3229 3230// ---------------------------------------------------------------------------- 3231 3232AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3233 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3234 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3235 mWaitTimeMs(UINT_MAX) 3236{ 3237 addOutputTrack(mainThread); 3238} 3239 3240AudioFlinger::DuplicatingThread::~DuplicatingThread() 3241{ 3242 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3243 mOutputTracks[i]->destroy(); 3244 } 3245} 3246 3247void AudioFlinger::DuplicatingThread::threadLoop_mix() 3248{ 3249 // mix buffers... 3250 if (outputsReady(outputTracks)) { 3251 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3252 } else { 3253 memset(mMixBuffer, 0, mixBufferSize); 3254 } 3255 sleepTime = 0; 3256 writeFrames = mFrameCount; 3257} 3258 3259void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3260{ 3261 if (sleepTime == 0) { 3262 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3263 sleepTime = activeSleepTime; 3264 } else { 3265 sleepTime = idleSleepTime; 3266 } 3267 } else if (mBytesWritten != 0) { 3268 // flush remaining overflow buffers in output tracks 3269 for (size_t i = 0; i < outputTracks.size(); i++) { 3270 if (outputTracks[i]->isActive()) { 3271 sleepTime = 0; 3272 writeFrames = 0; 3273 memset(mMixBuffer, 0, mixBufferSize); 3274 break; 3275 } 3276 } 3277 } 3278} 3279 3280void AudioFlinger::DuplicatingThread::threadLoop_write() 3281{ 3282 standbyTime = systemTime() + standbyDelay; 3283 for (size_t i = 0; i < outputTracks.size(); i++) { 3284 outputTracks[i]->write(mMixBuffer, writeFrames); 3285 } 3286 mBytesWritten += mixBufferSize; 3287} 3288 3289void AudioFlinger::DuplicatingThread::threadLoop_standby() 3290{ 3291 // DuplicatingThread implements standby by stopping all tracks 3292 for (size_t i = 0; i < outputTracks.size(); i++) { 3293 outputTracks[i]->stop(); 3294 } 3295} 3296 3297void AudioFlinger::DuplicatingThread::saveOutputTracks() 3298{ 3299 outputTracks = mOutputTracks; 3300} 3301 3302void AudioFlinger::DuplicatingThread::clearOutputTracks() 3303{ 3304 outputTracks.clear(); 3305} 3306 3307void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3308{ 3309 Mutex::Autolock _l(mLock); 3310 // FIXME explain this formula 3311 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3312 OutputTrack *outputTrack = new OutputTrack(thread, 3313 this, 3314 mSampleRate, 3315 mFormat, 3316 mChannelMask, 3317 frameCount); 3318 if (outputTrack->cblk() != NULL) { 3319 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3320 mOutputTracks.add(outputTrack); 3321 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3322 updateWaitTime_l(); 3323 } 3324} 3325 3326void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3327{ 3328 Mutex::Autolock _l(mLock); 3329 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3330 if (mOutputTracks[i]->thread() == thread) { 3331 mOutputTracks[i]->destroy(); 3332 mOutputTracks.removeAt(i); 3333 updateWaitTime_l(); 3334 return; 3335 } 3336 } 3337 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3338} 3339 3340// caller must hold mLock 3341void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3342{ 3343 mWaitTimeMs = UINT_MAX; 3344 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3345 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3346 if (strong != 0) { 3347 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3348 if (waitTimeMs < mWaitTimeMs) { 3349 mWaitTimeMs = waitTimeMs; 3350 } 3351 } 3352 } 3353} 3354 3355 3356bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3357{ 3358 for (size_t i = 0; i < outputTracks.size(); i++) { 3359 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3360 if (thread == 0) { 3361 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3362 return false; 3363 } 3364 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3365 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3366 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3367 return false; 3368 } 3369 } 3370 return true; 3371} 3372 3373uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3374{ 3375 return (mWaitTimeMs * 1000) / 2; 3376} 3377 3378void AudioFlinger::DuplicatingThread::cacheParameters_l() 3379{ 3380 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3381 updateWaitTime_l(); 3382 3383 MixerThread::cacheParameters_l(); 3384} 3385 3386// ---------------------------------------------------------------------------- 3387 3388// TrackBase constructor must be called with AudioFlinger::mLock held 3389AudioFlinger::ThreadBase::TrackBase::TrackBase( 3390 ThreadBase *thread, 3391 const sp<Client>& client, 3392 uint32_t sampleRate, 3393 audio_format_t format, 3394 uint32_t channelMask, 3395 int frameCount, 3396 const sp<IMemory>& sharedBuffer, 3397 int sessionId) 3398 : RefBase(), 3399 mThread(thread), 3400 mClient(client), 3401 mCblk(NULL), 3402 // mBuffer 3403 // mBufferEnd 3404 mFrameCount(0), 3405 mState(IDLE), 3406 mFormat(format), 3407 mStepServerFailed(false), 3408 mSessionId(sessionId) 3409 // mChannelCount 3410 // mChannelMask 3411{ 3412 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3413 3414 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3415 size_t size = sizeof(audio_track_cblk_t); 3416 uint8_t channelCount = popcount(channelMask); 3417 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3418 if (sharedBuffer == 0) { 3419 size += bufferSize; 3420 } 3421 3422 if (client != NULL) { 3423 mCblkMemory = client->heap()->allocate(size); 3424 if (mCblkMemory != 0) { 3425 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3426 if (mCblk != NULL) { // construct the shared structure in-place. 3427 new(mCblk) audio_track_cblk_t(); 3428 // clear all buffers 3429 mCblk->frameCount = frameCount; 3430 mCblk->sampleRate = sampleRate; 3431// uncomment the following lines to quickly test 32-bit wraparound 3432// mCblk->user = 0xffff0000; 3433// mCblk->server = 0xffff0000; 3434// mCblk->userBase = 0xffff0000; 3435// mCblk->serverBase = 0xffff0000; 3436 mChannelCount = channelCount; 3437 mChannelMask = channelMask; 3438 if (sharedBuffer == 0) { 3439 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3440 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3441 // Force underrun condition to avoid false underrun callback until first data is 3442 // written to buffer (other flags are cleared) 3443 mCblk->flags = CBLK_UNDERRUN_ON; 3444 } else { 3445 mBuffer = sharedBuffer->pointer(); 3446 } 3447 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3448 } 3449 } else { 3450 ALOGE("not enough memory for AudioTrack size=%u", size); 3451 client->heap()->dump("AudioTrack"); 3452 return; 3453 } 3454 } else { 3455 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3456 // construct the shared structure in-place. 3457 new(mCblk) audio_track_cblk_t(); 3458 // clear all buffers 3459 mCblk->frameCount = frameCount; 3460 mCblk->sampleRate = sampleRate; 3461// uncomment the following lines to quickly test 32-bit wraparound 3462// mCblk->user = 0xffff0000; 3463// mCblk->server = 0xffff0000; 3464// mCblk->userBase = 0xffff0000; 3465// mCblk->serverBase = 0xffff0000; 3466 mChannelCount = channelCount; 3467 mChannelMask = channelMask; 3468 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3469 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3470 // Force underrun condition to avoid false underrun callback until first data is 3471 // written to buffer (other flags are cleared) 3472 mCblk->flags = CBLK_UNDERRUN_ON; 3473 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3474 } 3475} 3476 3477AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3478{ 3479 if (mCblk != NULL) { 3480 if (mClient == 0) { 3481 delete mCblk; 3482 } else { 3483 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3484 } 3485 } 3486 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3487 if (mClient != 0) { 3488 // Client destructor must run with AudioFlinger mutex locked 3489 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3490 // If the client's reference count drops to zero, the associated destructor 3491 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3492 // relying on the automatic clear() at end of scope. 3493 mClient.clear(); 3494 } 3495} 3496 3497// AudioBufferProvider interface 3498// getNextBuffer() = 0; 3499// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3500void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3501{ 3502 buffer->raw = NULL; 3503 mFrameCount = buffer->frameCount; 3504 (void) step(); // ignore return value of step() 3505 buffer->frameCount = 0; 3506} 3507 3508bool AudioFlinger::ThreadBase::TrackBase::step() { 3509 bool result; 3510 audio_track_cblk_t* cblk = this->cblk(); 3511 3512 result = cblk->stepServer(mFrameCount); 3513 if (!result) { 3514 ALOGV("stepServer failed acquiring cblk mutex"); 3515 mStepServerFailed = true; 3516 } 3517 return result; 3518} 3519 3520void AudioFlinger::ThreadBase::TrackBase::reset() { 3521 audio_track_cblk_t* cblk = this->cblk(); 3522 3523 cblk->user = 0; 3524 cblk->server = 0; 3525 cblk->userBase = 0; 3526 cblk->serverBase = 0; 3527 mStepServerFailed = false; 3528 ALOGV("TrackBase::reset"); 3529} 3530 3531int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3532 return (int)mCblk->sampleRate; 3533} 3534 3535void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3536 audio_track_cblk_t* cblk = this->cblk(); 3537 size_t frameSize = cblk->frameSize; 3538 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3539 int8_t *bufferEnd = bufferStart + frames * frameSize; 3540 3541 // Check validity of returned pointer in case the track control block would have been corrupted. 3542 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3543 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3544 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3545 server %u, serverBase %u, user %u, userBase %u", 3546 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3547 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3548 return NULL; 3549 } 3550 3551 return bufferStart; 3552} 3553 3554status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3555{ 3556 mSyncEvents.add(event); 3557 return NO_ERROR; 3558} 3559 3560// ---------------------------------------------------------------------------- 3561 3562// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3563AudioFlinger::PlaybackThread::Track::Track( 3564 PlaybackThread *thread, 3565 const sp<Client>& client, 3566 audio_stream_type_t streamType, 3567 uint32_t sampleRate, 3568 audio_format_t format, 3569 uint32_t channelMask, 3570 int frameCount, 3571 const sp<IMemory>& sharedBuffer, 3572 int sessionId, 3573 IAudioFlinger::track_flags_t flags) 3574 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3575 mMute(false), 3576 // mFillingUpStatus ? 3577 // mRetryCount initialized later when needed 3578 mSharedBuffer(sharedBuffer), 3579 mStreamType(streamType), 3580 mName(-1), // see note below 3581 mMainBuffer(thread->mixBuffer()), 3582 mAuxBuffer(NULL), 3583 mAuxEffectId(0), mHasVolumeController(false), 3584 mPresentationCompleteFrames(0), 3585 mFlags(flags) 3586{ 3587 if (mCblk != NULL) { 3588 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3589 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3590 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3591 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3592 mName = thread->getTrackName_l(); 3593 if (mName < 0) { 3594 ALOGE("no more track names available"); 3595 } 3596 } 3597 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3598} 3599 3600AudioFlinger::PlaybackThread::Track::~Track() 3601{ 3602 ALOGV("PlaybackThread::Track destructor"); 3603 sp<ThreadBase> thread = mThread.promote(); 3604 if (thread != 0) { 3605 Mutex::Autolock _l(thread->mLock); 3606 mState = TERMINATED; 3607 } 3608} 3609 3610void AudioFlinger::PlaybackThread::Track::destroy() 3611{ 3612 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3613 // by removing it from mTracks vector, so there is a risk that this Tracks's 3614 // destructor is called. As the destructor needs to lock mLock, 3615 // we must acquire a strong reference on this Track before locking mLock 3616 // here so that the destructor is called only when exiting this function. 3617 // On the other hand, as long as Track::destroy() is only called by 3618 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3619 // this Track with its member mTrack. 3620 sp<Track> keep(this); 3621 { // scope for mLock 3622 sp<ThreadBase> thread = mThread.promote(); 3623 if (thread != 0) { 3624 if (!isOutputTrack()) { 3625 if (mState == ACTIVE || mState == RESUMING) { 3626 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3627 3628#ifdef ADD_BATTERY_DATA 3629 // to track the speaker usage 3630 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3631#endif 3632 } 3633 AudioSystem::releaseOutput(thread->id()); 3634 } 3635 Mutex::Autolock _l(thread->mLock); 3636 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3637 playbackThread->destroyTrack_l(this); 3638 } 3639 } 3640} 3641 3642void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3643{ 3644 uint32_t vlr = mCblk->getVolumeLR(); 3645 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", 3646 mName - AudioMixer::TRACK0, 3647 (mClient == 0) ? getpid_cached : mClient->pid(), 3648 mStreamType, 3649 mFormat, 3650 mChannelMask, 3651 mSessionId, 3652 mFrameCount, 3653 mState, 3654 mMute, 3655 mFillingUpStatus, 3656 mCblk->sampleRate, 3657 vlr & 0xFFFF, 3658 vlr >> 16, 3659 mCblk->server, 3660 mCblk->user, 3661 (int)mMainBuffer, 3662 (int)mAuxBuffer); 3663} 3664 3665// AudioBufferProvider interface 3666status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3667 AudioBufferProvider::Buffer* buffer, int64_t pts) 3668{ 3669 audio_track_cblk_t* cblk = this->cblk(); 3670 uint32_t framesReady; 3671 uint32_t framesReq = buffer->frameCount; 3672 3673 // Check if last stepServer failed, try to step now 3674 if (mStepServerFailed) { 3675 if (!step()) goto getNextBuffer_exit; 3676 ALOGV("stepServer recovered"); 3677 mStepServerFailed = false; 3678 } 3679 3680 framesReady = cblk->framesReady(); 3681 3682 if (CC_LIKELY(framesReady)) { 3683 uint32_t s = cblk->server; 3684 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3685 3686 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3687 if (framesReq > framesReady) { 3688 framesReq = framesReady; 3689 } 3690 if (framesReq > bufferEnd - s) { 3691 framesReq = bufferEnd - s; 3692 } 3693 3694 buffer->raw = getBuffer(s, framesReq); 3695 if (buffer->raw == NULL) goto getNextBuffer_exit; 3696 3697 buffer->frameCount = framesReq; 3698 return NO_ERROR; 3699 } 3700 3701getNextBuffer_exit: 3702 buffer->raw = NULL; 3703 buffer->frameCount = 0; 3704 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3705 return NOT_ENOUGH_DATA; 3706} 3707 3708uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3709 return mCblk->framesReady(); 3710} 3711 3712bool AudioFlinger::PlaybackThread::Track::isReady() const { 3713 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3714 3715 if (framesReady() >= mCblk->frameCount || 3716 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3717 mFillingUpStatus = FS_FILLED; 3718 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3719 return true; 3720 } 3721 return false; 3722} 3723 3724status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid, 3725 AudioSystem::sync_event_t event, 3726 int triggerSession) 3727{ 3728 status_t status = NO_ERROR; 3729 ALOGV("start(%d), calling pid %d session %d tid %d", 3730 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3731 // check for use case 2 with missing callback 3732 if (isFastTrack() && (mSharedBuffer == 0) && (tid == 0)) { 3733 ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied"); 3734 mFlags &= ~IAudioFlinger::TRACK_FAST; 3735 // FIXME the track must be invalidated and moved to another thread or 3736 // attached directly to the normal mixer now 3737 } 3738 sp<ThreadBase> thread = mThread.promote(); 3739 if (thread != 0) { 3740 Mutex::Autolock _l(thread->mLock); 3741 track_state state = mState; 3742 // here the track could be either new, or restarted 3743 // in both cases "unstop" the track 3744 if (mState == PAUSED) { 3745 mState = TrackBase::RESUMING; 3746 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3747 } else { 3748 mState = TrackBase::ACTIVE; 3749 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3750 } 3751 3752 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3753 thread->mLock.unlock(); 3754 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3755 thread->mLock.lock(); 3756 3757#ifdef ADD_BATTERY_DATA 3758 // to track the speaker usage 3759 if (status == NO_ERROR) { 3760 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3761 } 3762#endif 3763 } 3764 if (status == NO_ERROR) { 3765 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3766 playbackThread->addTrack_l(this); 3767 } else { 3768 mState = state; 3769 } 3770 } else { 3771 status = BAD_VALUE; 3772 } 3773 return status; 3774} 3775 3776void AudioFlinger::PlaybackThread::Track::stop() 3777{ 3778 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3779 sp<ThreadBase> thread = mThread.promote(); 3780 if (thread != 0) { 3781 Mutex::Autolock _l(thread->mLock); 3782 track_state state = mState; 3783 if (mState > STOPPED) { 3784 mState = STOPPED; 3785 // If the track is not active (PAUSED and buffers full), flush buffers 3786 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3787 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3788 reset(); 3789 } 3790 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3791 } 3792 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3793 thread->mLock.unlock(); 3794 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3795 thread->mLock.lock(); 3796 3797#ifdef ADD_BATTERY_DATA 3798 // to track the speaker usage 3799 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3800#endif 3801 } 3802 } 3803} 3804 3805void AudioFlinger::PlaybackThread::Track::pause() 3806{ 3807 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3808 sp<ThreadBase> thread = mThread.promote(); 3809 if (thread != 0) { 3810 Mutex::Autolock _l(thread->mLock); 3811 if (mState == ACTIVE || mState == RESUMING) { 3812 mState = PAUSING; 3813 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3814 if (!isOutputTrack()) { 3815 thread->mLock.unlock(); 3816 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3817 thread->mLock.lock(); 3818 3819#ifdef ADD_BATTERY_DATA 3820 // to track the speaker usage 3821 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3822#endif 3823 } 3824 } 3825 } 3826} 3827 3828void AudioFlinger::PlaybackThread::Track::flush() 3829{ 3830 ALOGV("flush(%d)", mName); 3831 sp<ThreadBase> thread = mThread.promote(); 3832 if (thread != 0) { 3833 Mutex::Autolock _l(thread->mLock); 3834 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3835 return; 3836 } 3837 // No point remaining in PAUSED state after a flush => go to 3838 // STOPPED state 3839 mState = STOPPED; 3840 3841 // do not reset the track if it is still in the process of being stopped or paused. 3842 // this will be done by prepareTracks_l() when the track is stopped. 3843 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3844 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3845 reset(); 3846 } 3847 } 3848} 3849 3850void AudioFlinger::PlaybackThread::Track::reset() 3851{ 3852 // Do not reset twice to avoid discarding data written just after a flush and before 3853 // the audioflinger thread detects the track is stopped. 3854 if (!mResetDone) { 3855 TrackBase::reset(); 3856 // Force underrun condition to avoid false underrun callback until first data is 3857 // written to buffer 3858 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3859 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3860 mFillingUpStatus = FS_FILLING; 3861 mResetDone = true; 3862 mPresentationCompleteFrames = 0; 3863 } 3864} 3865 3866void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3867{ 3868 mMute = muted; 3869} 3870 3871status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3872{ 3873 status_t status = DEAD_OBJECT; 3874 sp<ThreadBase> thread = mThread.promote(); 3875 if (thread != 0) { 3876 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3877 status = playbackThread->attachAuxEffect(this, EffectId); 3878 } 3879 return status; 3880} 3881 3882void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3883{ 3884 mAuxEffectId = EffectId; 3885 mAuxBuffer = buffer; 3886} 3887 3888bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 3889 size_t audioHalFrames) 3890{ 3891 // a track is considered presented when the total number of frames written to audio HAL 3892 // corresponds to the number of frames written when presentationComplete() is called for the 3893 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 3894 if (mPresentationCompleteFrames == 0) { 3895 mPresentationCompleteFrames = framesWritten + audioHalFrames; 3896 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 3897 mPresentationCompleteFrames, audioHalFrames); 3898 } 3899 if (framesWritten >= mPresentationCompleteFrames) { 3900 ALOGV("presentationComplete() session %d complete: framesWritten %d", 3901 mSessionId, framesWritten); 3902 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 3903 mPresentationCompleteFrames = 0; 3904 return true; 3905 } 3906 return false; 3907} 3908 3909void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 3910{ 3911 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 3912 if (mSyncEvents[i]->type() == type) { 3913 mSyncEvents[i]->trigger(); 3914 mSyncEvents.removeAt(i); 3915 i--; 3916 } 3917 } 3918} 3919 3920 3921// timed audio tracks 3922 3923sp<AudioFlinger::PlaybackThread::TimedTrack> 3924AudioFlinger::PlaybackThread::TimedTrack::create( 3925 PlaybackThread *thread, 3926 const sp<Client>& client, 3927 audio_stream_type_t streamType, 3928 uint32_t sampleRate, 3929 audio_format_t format, 3930 uint32_t channelMask, 3931 int frameCount, 3932 const sp<IMemory>& sharedBuffer, 3933 int sessionId) { 3934 if (!client->reserveTimedTrack()) 3935 return NULL; 3936 3937 return new TimedTrack( 3938 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3939 sharedBuffer, sessionId); 3940} 3941 3942AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3943 PlaybackThread *thread, 3944 const sp<Client>& client, 3945 audio_stream_type_t streamType, 3946 uint32_t sampleRate, 3947 audio_format_t format, 3948 uint32_t channelMask, 3949 int frameCount, 3950 const sp<IMemory>& sharedBuffer, 3951 int sessionId) 3952 : Track(thread, client, streamType, sampleRate, format, channelMask, 3953 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 3954 mTimedSilenceBuffer(NULL), 3955 mTimedSilenceBufferSize(0), 3956 mTimedAudioOutputOnTime(false), 3957 mMediaTimeTransformValid(false) 3958{ 3959 LocalClock lc; 3960 mLocalTimeFreq = lc.getLocalFreq(); 3961 3962 mLocalTimeToSampleTransform.a_zero = 0; 3963 mLocalTimeToSampleTransform.b_zero = 0; 3964 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3965 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3966 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3967 &mLocalTimeToSampleTransform.a_to_b_denom); 3968} 3969 3970AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3971 mClient->releaseTimedTrack(); 3972 delete [] mTimedSilenceBuffer; 3973} 3974 3975status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3976 size_t size, sp<IMemory>* buffer) { 3977 3978 Mutex::Autolock _l(mTimedBufferQueueLock); 3979 3980 trimTimedBufferQueue_l(); 3981 3982 // lazily initialize the shared memory heap for timed buffers 3983 if (mTimedMemoryDealer == NULL) { 3984 const int kTimedBufferHeapSize = 512 << 10; 3985 3986 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3987 "AudioFlingerTimed"); 3988 if (mTimedMemoryDealer == NULL) 3989 return NO_MEMORY; 3990 } 3991 3992 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3993 if (newBuffer == NULL) { 3994 newBuffer = mTimedMemoryDealer->allocate(size); 3995 if (newBuffer == NULL) 3996 return NO_MEMORY; 3997 } 3998 3999 *buffer = newBuffer; 4000 return NO_ERROR; 4001} 4002 4003// caller must hold mTimedBufferQueueLock 4004void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4005 int64_t mediaTimeNow; 4006 { 4007 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4008 if (!mMediaTimeTransformValid) 4009 return; 4010 4011 int64_t targetTimeNow; 4012 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4013 ? mCCHelper.getCommonTime(&targetTimeNow) 4014 : mCCHelper.getLocalTime(&targetTimeNow); 4015 4016 if (OK != res) 4017 return; 4018 4019 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4020 &mediaTimeNow)) { 4021 return; 4022 } 4023 } 4024 4025 size_t trimIndex; 4026 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 4027 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 4028 break; 4029 } 4030 4031 if (trimIndex) { 4032 mTimedBufferQueue.removeItemsAt(0, trimIndex); 4033 } 4034} 4035 4036status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4037 const sp<IMemory>& buffer, int64_t pts) { 4038 4039 { 4040 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4041 if (!mMediaTimeTransformValid) 4042 return INVALID_OPERATION; 4043 } 4044 4045 Mutex::Autolock _l(mTimedBufferQueueLock); 4046 4047 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4048 4049 return NO_ERROR; 4050} 4051 4052status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4053 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4054 4055 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 4056 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4057 target); 4058 4059 if (!(target == TimedAudioTrack::LOCAL_TIME || 4060 target == TimedAudioTrack::COMMON_TIME)) { 4061 return BAD_VALUE; 4062 } 4063 4064 Mutex::Autolock lock(mMediaTimeTransformLock); 4065 mMediaTimeTransform = xform; 4066 mMediaTimeTransformTarget = target; 4067 mMediaTimeTransformValid = true; 4068 4069 return NO_ERROR; 4070} 4071 4072#define min(a, b) ((a) < (b) ? (a) : (b)) 4073 4074// implementation of getNextBuffer for tracks whose buffers have timestamps 4075status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4076 AudioBufferProvider::Buffer* buffer, int64_t pts) 4077{ 4078 if (pts == AudioBufferProvider::kInvalidPTS) { 4079 buffer->raw = 0; 4080 buffer->frameCount = 0; 4081 return INVALID_OPERATION; 4082 } 4083 4084 Mutex::Autolock _l(mTimedBufferQueueLock); 4085 4086 while (true) { 4087 4088 // if we have no timed buffers, then fail 4089 if (mTimedBufferQueue.isEmpty()) { 4090 buffer->raw = 0; 4091 buffer->frameCount = 0; 4092 return NOT_ENOUGH_DATA; 4093 } 4094 4095 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4096 4097 // calculate the PTS of the head of the timed buffer queue expressed in 4098 // local time 4099 int64_t headLocalPTS; 4100 { 4101 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4102 4103 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4104 4105 if (mMediaTimeTransform.a_to_b_denom == 0) { 4106 // the transform represents a pause, so yield silence 4107 timedYieldSilence(buffer->frameCount, buffer); 4108 return NO_ERROR; 4109 } 4110 4111 int64_t transformedPTS; 4112 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4113 &transformedPTS)) { 4114 // the transform failed. this shouldn't happen, but if it does 4115 // then just drop this buffer 4116 ALOGW("timedGetNextBuffer transform failed"); 4117 buffer->raw = 0; 4118 buffer->frameCount = 0; 4119 mTimedBufferQueue.removeAt(0); 4120 return NO_ERROR; 4121 } 4122 4123 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4124 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4125 &headLocalPTS)) { 4126 buffer->raw = 0; 4127 buffer->frameCount = 0; 4128 return INVALID_OPERATION; 4129 } 4130 } else { 4131 headLocalPTS = transformedPTS; 4132 } 4133 } 4134 4135 // adjust the head buffer's PTS to reflect the portion of the head buffer 4136 // that has already been consumed 4137 int64_t effectivePTS = headLocalPTS + 4138 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4139 4140 // Calculate the delta in samples between the head of the input buffer 4141 // queue and the start of the next output buffer that will be written. 4142 // If the transformation fails because of over or underflow, it means 4143 // that the sample's position in the output stream is so far out of 4144 // whack that it should just be dropped. 4145 int64_t sampleDelta; 4146 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4147 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4148 mTimedBufferQueue.removeAt(0); 4149 continue; 4150 } 4151 if (!mLocalTimeToSampleTransform.doForwardTransform( 4152 (effectivePTS - pts) << 32, &sampleDelta)) { 4153 ALOGV("*** too late during sample rate transform: dropped buffer"); 4154 mTimedBufferQueue.removeAt(0); 4155 continue; 4156 } 4157 4158 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4159 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4160 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4161 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4162 4163 // if the delta between the ideal placement for the next input sample and 4164 // the current output position is within this threshold, then we will 4165 // concatenate the next input samples to the previous output 4166 const int64_t kSampleContinuityThreshold = 4167 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4168 4169 // if this is the first buffer of audio that we're emitting from this track 4170 // then it should be almost exactly on time. 4171 const int64_t kSampleStartupThreshold = 1LL << 32; 4172 4173 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4174 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4175 // the next input is close enough to being on time, so concatenate it 4176 // with the last output 4177 timedYieldSamples(buffer); 4178 4179 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4180 return NO_ERROR; 4181 } else if (sampleDelta > 0) { 4182 // the gap between the current output position and the proper start of 4183 // the next input sample is too big, so fill it with silence 4184 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4185 4186 timedYieldSilence(framesUntilNextInput, buffer); 4187 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4188 return NO_ERROR; 4189 } else { 4190 // the next input sample is late 4191 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4192 size_t onTimeSamplePosition = 4193 head.position() + lateFrames * mCblk->frameSize; 4194 4195 if (onTimeSamplePosition > head.buffer()->size()) { 4196 // all the remaining samples in the head are too late, so 4197 // drop it and move on 4198 ALOGV("*** too late: dropped buffer"); 4199 mTimedBufferQueue.removeAt(0); 4200 continue; 4201 } else { 4202 // skip over the late samples 4203 head.setPosition(onTimeSamplePosition); 4204 4205 // yield the available samples 4206 timedYieldSamples(buffer); 4207 4208 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4209 return NO_ERROR; 4210 } 4211 } 4212 } 4213} 4214 4215// Yield samples from the timed buffer queue head up to the given output 4216// buffer's capacity. 4217// 4218// Caller must hold mTimedBufferQueueLock 4219void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4220 AudioBufferProvider::Buffer* buffer) { 4221 4222 const TimedBuffer& head = mTimedBufferQueue[0]; 4223 4224 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4225 head.position()); 4226 4227 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4228 mCblk->frameSize); 4229 size_t framesRequested = buffer->frameCount; 4230 buffer->frameCount = min(framesLeftInHead, framesRequested); 4231 4232 mTimedAudioOutputOnTime = true; 4233} 4234 4235// Yield samples of silence up to the given output buffer's capacity 4236// 4237// Caller must hold mTimedBufferQueueLock 4238void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4239 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4240 4241 // lazily allocate a buffer filled with silence 4242 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4243 delete [] mTimedSilenceBuffer; 4244 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4245 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4246 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4247 } 4248 4249 buffer->raw = mTimedSilenceBuffer; 4250 size_t framesRequested = buffer->frameCount; 4251 buffer->frameCount = min(numFrames, framesRequested); 4252 4253 mTimedAudioOutputOnTime = false; 4254} 4255 4256// AudioBufferProvider interface 4257void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4258 AudioBufferProvider::Buffer* buffer) { 4259 4260 Mutex::Autolock _l(mTimedBufferQueueLock); 4261 4262 // If the buffer which was just released is part of the buffer at the head 4263 // of the queue, be sure to update the amt of the buffer which has been 4264 // consumed. If the buffer being returned is not part of the head of the 4265 // queue, its either because the buffer is part of the silence buffer, or 4266 // because the head of the timed queue was trimmed after the mixer called 4267 // getNextBuffer but before the mixer called releaseBuffer. 4268 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4269 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4270 4271 void* start = head.buffer()->pointer(); 4272 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4273 4274 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4275 head.setPosition(head.position() + 4276 (buffer->frameCount * mCblk->frameSize)); 4277 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4278 mTimedBufferQueue.removeAt(0); 4279 } 4280 } 4281 } 4282 4283 buffer->raw = 0; 4284 buffer->frameCount = 0; 4285} 4286 4287uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4288 Mutex::Autolock _l(mTimedBufferQueueLock); 4289 4290 uint32_t frames = 0; 4291 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4292 const TimedBuffer& tb = mTimedBufferQueue[i]; 4293 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4294 } 4295 4296 return frames; 4297} 4298 4299AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4300 : mPTS(0), mPosition(0) {} 4301 4302AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4303 const sp<IMemory>& buffer, int64_t pts) 4304 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4305 4306// ---------------------------------------------------------------------------- 4307 4308// RecordTrack constructor must be called with AudioFlinger::mLock held 4309AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4310 RecordThread *thread, 4311 const sp<Client>& client, 4312 uint32_t sampleRate, 4313 audio_format_t format, 4314 uint32_t channelMask, 4315 int frameCount, 4316 int sessionId) 4317 : TrackBase(thread, client, sampleRate, format, 4318 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4319 mOverflow(false) 4320{ 4321 if (mCblk != NULL) { 4322 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4323 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4324 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4325 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4326 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4327 } else { 4328 mCblk->frameSize = sizeof(int8_t); 4329 } 4330 } 4331} 4332 4333AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4334{ 4335 sp<ThreadBase> thread = mThread.promote(); 4336 if (thread != 0) { 4337 AudioSystem::releaseInput(thread->id()); 4338 } 4339} 4340 4341// AudioBufferProvider interface 4342status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4343{ 4344 audio_track_cblk_t* cblk = this->cblk(); 4345 uint32_t framesAvail; 4346 uint32_t framesReq = buffer->frameCount; 4347 4348 // Check if last stepServer failed, try to step now 4349 if (mStepServerFailed) { 4350 if (!step()) goto getNextBuffer_exit; 4351 ALOGV("stepServer recovered"); 4352 mStepServerFailed = false; 4353 } 4354 4355 framesAvail = cblk->framesAvailable_l(); 4356 4357 if (CC_LIKELY(framesAvail)) { 4358 uint32_t s = cblk->server; 4359 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4360 4361 if (framesReq > framesAvail) { 4362 framesReq = framesAvail; 4363 } 4364 if (framesReq > bufferEnd - s) { 4365 framesReq = bufferEnd - s; 4366 } 4367 4368 buffer->raw = getBuffer(s, framesReq); 4369 if (buffer->raw == NULL) goto getNextBuffer_exit; 4370 4371 buffer->frameCount = framesReq; 4372 return NO_ERROR; 4373 } 4374 4375getNextBuffer_exit: 4376 buffer->raw = NULL; 4377 buffer->frameCount = 0; 4378 return NOT_ENOUGH_DATA; 4379} 4380 4381status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid, 4382 AudioSystem::sync_event_t event, 4383 int triggerSession) 4384{ 4385 sp<ThreadBase> thread = mThread.promote(); 4386 if (thread != 0) { 4387 RecordThread *recordThread = (RecordThread *)thread.get(); 4388 return recordThread->start(this, tid, event, triggerSession); 4389 } else { 4390 return BAD_VALUE; 4391 } 4392} 4393 4394void AudioFlinger::RecordThread::RecordTrack::stop() 4395{ 4396 sp<ThreadBase> thread = mThread.promote(); 4397 if (thread != 0) { 4398 RecordThread *recordThread = (RecordThread *)thread.get(); 4399 recordThread->stop(this); 4400 TrackBase::reset(); 4401 // Force overrun condition to avoid false overrun callback until first data is 4402 // read from buffer 4403 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4404 } 4405} 4406 4407void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4408{ 4409 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4410 (mClient == 0) ? getpid_cached : mClient->pid(), 4411 mFormat, 4412 mChannelMask, 4413 mSessionId, 4414 mFrameCount, 4415 mState, 4416 mCblk->sampleRate, 4417 mCblk->server, 4418 mCblk->user); 4419} 4420 4421 4422// ---------------------------------------------------------------------------- 4423 4424AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4425 PlaybackThread *playbackThread, 4426 DuplicatingThread *sourceThread, 4427 uint32_t sampleRate, 4428 audio_format_t format, 4429 uint32_t channelMask, 4430 int frameCount) 4431 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 4432 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 4433 mActive(false), mSourceThread(sourceThread) 4434{ 4435 4436 if (mCblk != NULL) { 4437 mCblk->flags |= CBLK_DIRECTION_OUT; 4438 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4439 mOutBuffer.frameCount = 0; 4440 playbackThread->mTracks.add(this); 4441 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4442 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4443 mCblk, mBuffer, mCblk->buffers, 4444 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4445 } else { 4446 ALOGW("Error creating output track on thread %p", playbackThread); 4447 } 4448} 4449 4450AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4451{ 4452 clearBufferQueue(); 4453} 4454 4455status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid, 4456 AudioSystem::sync_event_t event, 4457 int triggerSession) 4458{ 4459 status_t status = Track::start(tid, event, triggerSession); 4460 if (status != NO_ERROR) { 4461 return status; 4462 } 4463 4464 mActive = true; 4465 mRetryCount = 127; 4466 return status; 4467} 4468 4469void AudioFlinger::PlaybackThread::OutputTrack::stop() 4470{ 4471 Track::stop(); 4472 clearBufferQueue(); 4473 mOutBuffer.frameCount = 0; 4474 mActive = false; 4475} 4476 4477bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4478{ 4479 Buffer *pInBuffer; 4480 Buffer inBuffer; 4481 uint32_t channelCount = mChannelCount; 4482 bool outputBufferFull = false; 4483 inBuffer.frameCount = frames; 4484 inBuffer.i16 = data; 4485 4486 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4487 4488 if (!mActive && frames != 0) { 4489 start(0); 4490 sp<ThreadBase> thread = mThread.promote(); 4491 if (thread != 0) { 4492 MixerThread *mixerThread = (MixerThread *)thread.get(); 4493 if (mCblk->frameCount > frames){ 4494 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4495 uint32_t startFrames = (mCblk->frameCount - frames); 4496 pInBuffer = new Buffer; 4497 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4498 pInBuffer->frameCount = startFrames; 4499 pInBuffer->i16 = pInBuffer->mBuffer; 4500 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4501 mBufferQueue.add(pInBuffer); 4502 } else { 4503 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4504 } 4505 } 4506 } 4507 } 4508 4509 while (waitTimeLeftMs) { 4510 // First write pending buffers, then new data 4511 if (mBufferQueue.size()) { 4512 pInBuffer = mBufferQueue.itemAt(0); 4513 } else { 4514 pInBuffer = &inBuffer; 4515 } 4516 4517 if (pInBuffer->frameCount == 0) { 4518 break; 4519 } 4520 4521 if (mOutBuffer.frameCount == 0) { 4522 mOutBuffer.frameCount = pInBuffer->frameCount; 4523 nsecs_t startTime = systemTime(); 4524 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4525 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4526 outputBufferFull = true; 4527 break; 4528 } 4529 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4530 if (waitTimeLeftMs >= waitTimeMs) { 4531 waitTimeLeftMs -= waitTimeMs; 4532 } else { 4533 waitTimeLeftMs = 0; 4534 } 4535 } 4536 4537 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4538 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4539 mCblk->stepUser(outFrames); 4540 pInBuffer->frameCount -= outFrames; 4541 pInBuffer->i16 += outFrames * channelCount; 4542 mOutBuffer.frameCount -= outFrames; 4543 mOutBuffer.i16 += outFrames * channelCount; 4544 4545 if (pInBuffer->frameCount == 0) { 4546 if (mBufferQueue.size()) { 4547 mBufferQueue.removeAt(0); 4548 delete [] pInBuffer->mBuffer; 4549 delete pInBuffer; 4550 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4551 } else { 4552 break; 4553 } 4554 } 4555 } 4556 4557 // If we could not write all frames, allocate a buffer and queue it for next time. 4558 if (inBuffer.frameCount) { 4559 sp<ThreadBase> thread = mThread.promote(); 4560 if (thread != 0 && !thread->standby()) { 4561 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4562 pInBuffer = new Buffer; 4563 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4564 pInBuffer->frameCount = inBuffer.frameCount; 4565 pInBuffer->i16 = pInBuffer->mBuffer; 4566 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4567 mBufferQueue.add(pInBuffer); 4568 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4569 } else { 4570 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4571 } 4572 } 4573 } 4574 4575 // Calling write() with a 0 length buffer, means that no more data will be written: 4576 // If no more buffers are pending, fill output track buffer to make sure it is started 4577 // by output mixer. 4578 if (frames == 0 && mBufferQueue.size() == 0) { 4579 if (mCblk->user < mCblk->frameCount) { 4580 frames = mCblk->frameCount - mCblk->user; 4581 pInBuffer = new Buffer; 4582 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4583 pInBuffer->frameCount = frames; 4584 pInBuffer->i16 = pInBuffer->mBuffer; 4585 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4586 mBufferQueue.add(pInBuffer); 4587 } else if (mActive) { 4588 stop(); 4589 } 4590 } 4591 4592 return outputBufferFull; 4593} 4594 4595status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4596{ 4597 int active; 4598 status_t result; 4599 audio_track_cblk_t* cblk = mCblk; 4600 uint32_t framesReq = buffer->frameCount; 4601 4602// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4603 buffer->frameCount = 0; 4604 4605 uint32_t framesAvail = cblk->framesAvailable(); 4606 4607 4608 if (framesAvail == 0) { 4609 Mutex::Autolock _l(cblk->lock); 4610 goto start_loop_here; 4611 while (framesAvail == 0) { 4612 active = mActive; 4613 if (CC_UNLIKELY(!active)) { 4614 ALOGV("Not active and NO_MORE_BUFFERS"); 4615 return NO_MORE_BUFFERS; 4616 } 4617 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4618 if (result != NO_ERROR) { 4619 return NO_MORE_BUFFERS; 4620 } 4621 // read the server count again 4622 start_loop_here: 4623 framesAvail = cblk->framesAvailable_l(); 4624 } 4625 } 4626 4627// if (framesAvail < framesReq) { 4628// return NO_MORE_BUFFERS; 4629// } 4630 4631 if (framesReq > framesAvail) { 4632 framesReq = framesAvail; 4633 } 4634 4635 uint32_t u = cblk->user; 4636 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4637 4638 if (framesReq > bufferEnd - u) { 4639 framesReq = bufferEnd - u; 4640 } 4641 4642 buffer->frameCount = framesReq; 4643 buffer->raw = (void *)cblk->buffer(u); 4644 return NO_ERROR; 4645} 4646 4647 4648void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4649{ 4650 size_t size = mBufferQueue.size(); 4651 4652 for (size_t i = 0; i < size; i++) { 4653 Buffer *pBuffer = mBufferQueue.itemAt(i); 4654 delete [] pBuffer->mBuffer; 4655 delete pBuffer; 4656 } 4657 mBufferQueue.clear(); 4658} 4659 4660// ---------------------------------------------------------------------------- 4661 4662AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4663 : RefBase(), 4664 mAudioFlinger(audioFlinger), 4665 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4666 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4667 mPid(pid), 4668 mTimedTrackCount(0) 4669{ 4670 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4671} 4672 4673// Client destructor must be called with AudioFlinger::mLock held 4674AudioFlinger::Client::~Client() 4675{ 4676 mAudioFlinger->removeClient_l(mPid); 4677} 4678 4679sp<MemoryDealer> AudioFlinger::Client::heap() const 4680{ 4681 return mMemoryDealer; 4682} 4683 4684// Reserve one of the limited slots for a timed audio track associated 4685// with this client 4686bool AudioFlinger::Client::reserveTimedTrack() 4687{ 4688 const int kMaxTimedTracksPerClient = 4; 4689 4690 Mutex::Autolock _l(mTimedTrackLock); 4691 4692 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4693 ALOGW("can not create timed track - pid %d has exceeded the limit", 4694 mPid); 4695 return false; 4696 } 4697 4698 mTimedTrackCount++; 4699 return true; 4700} 4701 4702// Release a slot for a timed audio track 4703void AudioFlinger::Client::releaseTimedTrack() 4704{ 4705 Mutex::Autolock _l(mTimedTrackLock); 4706 mTimedTrackCount--; 4707} 4708 4709// ---------------------------------------------------------------------------- 4710 4711AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4712 const sp<IAudioFlingerClient>& client, 4713 pid_t pid) 4714 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4715{ 4716} 4717 4718AudioFlinger::NotificationClient::~NotificationClient() 4719{ 4720} 4721 4722void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4723{ 4724 sp<NotificationClient> keep(this); 4725 mAudioFlinger->removeNotificationClient(mPid); 4726} 4727 4728// ---------------------------------------------------------------------------- 4729 4730AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4731 : BnAudioTrack(), 4732 mTrack(track) 4733{ 4734} 4735 4736AudioFlinger::TrackHandle::~TrackHandle() { 4737 // just stop the track on deletion, associated resources 4738 // will be freed from the main thread once all pending buffers have 4739 // been played. Unless it's not in the active track list, in which 4740 // case we free everything now... 4741 mTrack->destroy(); 4742} 4743 4744sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4745 return mTrack->getCblk(); 4746} 4747 4748status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4749 return mTrack->start(tid); 4750} 4751 4752void AudioFlinger::TrackHandle::stop() { 4753 mTrack->stop(); 4754} 4755 4756void AudioFlinger::TrackHandle::flush() { 4757 mTrack->flush(); 4758} 4759 4760void AudioFlinger::TrackHandle::mute(bool e) { 4761 mTrack->mute(e); 4762} 4763 4764void AudioFlinger::TrackHandle::pause() { 4765 mTrack->pause(); 4766} 4767 4768status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4769{ 4770 return mTrack->attachAuxEffect(EffectId); 4771} 4772 4773status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4774 sp<IMemory>* buffer) { 4775 if (!mTrack->isTimedTrack()) 4776 return INVALID_OPERATION; 4777 4778 PlaybackThread::TimedTrack* tt = 4779 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4780 return tt->allocateTimedBuffer(size, buffer); 4781} 4782 4783status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4784 int64_t pts) { 4785 if (!mTrack->isTimedTrack()) 4786 return INVALID_OPERATION; 4787 4788 PlaybackThread::TimedTrack* tt = 4789 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4790 return tt->queueTimedBuffer(buffer, pts); 4791} 4792 4793status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4794 const LinearTransform& xform, int target) { 4795 4796 if (!mTrack->isTimedTrack()) 4797 return INVALID_OPERATION; 4798 4799 PlaybackThread::TimedTrack* tt = 4800 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4801 return tt->setMediaTimeTransform( 4802 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4803} 4804 4805status_t AudioFlinger::TrackHandle::onTransact( 4806 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4807{ 4808 return BnAudioTrack::onTransact(code, data, reply, flags); 4809} 4810 4811// ---------------------------------------------------------------------------- 4812 4813sp<IAudioRecord> AudioFlinger::openRecord( 4814 pid_t pid, 4815 audio_io_handle_t input, 4816 uint32_t sampleRate, 4817 audio_format_t format, 4818 uint32_t channelMask, 4819 int frameCount, 4820 IAudioFlinger::track_flags_t flags, 4821 int *sessionId, 4822 status_t *status) 4823{ 4824 sp<RecordThread::RecordTrack> recordTrack; 4825 sp<RecordHandle> recordHandle; 4826 sp<Client> client; 4827 status_t lStatus; 4828 RecordThread *thread; 4829 size_t inFrameCount; 4830 int lSessionId; 4831 4832 // check calling permissions 4833 if (!recordingAllowed()) { 4834 lStatus = PERMISSION_DENIED; 4835 goto Exit; 4836 } 4837 4838 // add client to list 4839 { // scope for mLock 4840 Mutex::Autolock _l(mLock); 4841 thread = checkRecordThread_l(input); 4842 if (thread == NULL) { 4843 lStatus = BAD_VALUE; 4844 goto Exit; 4845 } 4846 4847 client = registerPid_l(pid); 4848 4849 // If no audio session id is provided, create one here 4850 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4851 lSessionId = *sessionId; 4852 } else { 4853 lSessionId = nextUniqueId(); 4854 if (sessionId != NULL) { 4855 *sessionId = lSessionId; 4856 } 4857 } 4858 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4859 recordTrack = thread->createRecordTrack_l(client, 4860 sampleRate, 4861 format, 4862 channelMask, 4863 frameCount, 4864 lSessionId, 4865 &lStatus); 4866 } 4867 if (lStatus != NO_ERROR) { 4868 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4869 // destructor is called by the TrackBase destructor with mLock held 4870 client.clear(); 4871 recordTrack.clear(); 4872 goto Exit; 4873 } 4874 4875 // return to handle to client 4876 recordHandle = new RecordHandle(recordTrack); 4877 lStatus = NO_ERROR; 4878 4879Exit: 4880 if (status) { 4881 *status = lStatus; 4882 } 4883 return recordHandle; 4884} 4885 4886// ---------------------------------------------------------------------------- 4887 4888AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4889 : BnAudioRecord(), 4890 mRecordTrack(recordTrack) 4891{ 4892} 4893 4894AudioFlinger::RecordHandle::~RecordHandle() { 4895 stop(); 4896} 4897 4898sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4899 return mRecordTrack->getCblk(); 4900} 4901 4902status_t AudioFlinger::RecordHandle::start(pid_t tid, int event, int triggerSession) { 4903 ALOGV("RecordHandle::start()"); 4904 return mRecordTrack->start(tid, (AudioSystem::sync_event_t)event, triggerSession); 4905} 4906 4907void AudioFlinger::RecordHandle::stop() { 4908 ALOGV("RecordHandle::stop()"); 4909 mRecordTrack->stop(); 4910} 4911 4912status_t AudioFlinger::RecordHandle::onTransact( 4913 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4914{ 4915 return BnAudioRecord::onTransact(code, data, reply, flags); 4916} 4917 4918// ---------------------------------------------------------------------------- 4919 4920AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4921 AudioStreamIn *input, 4922 uint32_t sampleRate, 4923 uint32_t channels, 4924 audio_io_handle_t id, 4925 uint32_t device) : 4926 ThreadBase(audioFlinger, id, device, RECORD), 4927 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4928 // mRsmpInIndex and mInputBytes set by readInputParameters() 4929 mReqChannelCount(popcount(channels)), 4930 mReqSampleRate(sampleRate) 4931 // mBytesRead is only meaningful while active, and so is cleared in start() 4932 // (but might be better to also clear here for dump?) 4933{ 4934 snprintf(mName, kNameLength, "AudioIn_%X", id); 4935 4936 readInputParameters(); 4937} 4938 4939 4940AudioFlinger::RecordThread::~RecordThread() 4941{ 4942 delete[] mRsmpInBuffer; 4943 delete mResampler; 4944 delete[] mRsmpOutBuffer; 4945} 4946 4947void AudioFlinger::RecordThread::onFirstRef() 4948{ 4949 run(mName, PRIORITY_URGENT_AUDIO); 4950} 4951 4952status_t AudioFlinger::RecordThread::readyToRun() 4953{ 4954 status_t status = initCheck(); 4955 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4956 return status; 4957} 4958 4959bool AudioFlinger::RecordThread::threadLoop() 4960{ 4961 AudioBufferProvider::Buffer buffer; 4962 sp<RecordTrack> activeTrack; 4963 Vector< sp<EffectChain> > effectChains; 4964 4965 nsecs_t lastWarning = 0; 4966 4967 acquireWakeLock(); 4968 4969 // start recording 4970 while (!exitPending()) { 4971 4972 processConfigEvents(); 4973 4974 { // scope for mLock 4975 Mutex::Autolock _l(mLock); 4976 checkForNewParameters_l(); 4977 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4978 if (!mStandby) { 4979 mInput->stream->common.standby(&mInput->stream->common); 4980 mStandby = true; 4981 } 4982 4983 if (exitPending()) break; 4984 4985 releaseWakeLock_l(); 4986 ALOGV("RecordThread: loop stopping"); 4987 // go to sleep 4988 mWaitWorkCV.wait(mLock); 4989 ALOGV("RecordThread: loop starting"); 4990 acquireWakeLock_l(); 4991 continue; 4992 } 4993 if (mActiveTrack != 0) { 4994 if (mActiveTrack->mState == TrackBase::PAUSING) { 4995 if (!mStandby) { 4996 mInput->stream->common.standby(&mInput->stream->common); 4997 mStandby = true; 4998 } 4999 mActiveTrack.clear(); 5000 mStartStopCond.broadcast(); 5001 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5002 if (mReqChannelCount != mActiveTrack->channelCount()) { 5003 mActiveTrack.clear(); 5004 mStartStopCond.broadcast(); 5005 } else if (mBytesRead != 0) { 5006 // record start succeeds only if first read from audio input 5007 // succeeds 5008 if (mBytesRead > 0) { 5009 mActiveTrack->mState = TrackBase::ACTIVE; 5010 } else { 5011 mActiveTrack.clear(); 5012 } 5013 mStartStopCond.broadcast(); 5014 } 5015 mStandby = false; 5016 } 5017 } 5018 lockEffectChains_l(effectChains); 5019 } 5020 5021 if (mActiveTrack != 0) { 5022 if (mActiveTrack->mState != TrackBase::ACTIVE && 5023 mActiveTrack->mState != TrackBase::RESUMING) { 5024 unlockEffectChains(effectChains); 5025 usleep(kRecordThreadSleepUs); 5026 continue; 5027 } 5028 for (size_t i = 0; i < effectChains.size(); i ++) { 5029 effectChains[i]->process_l(); 5030 } 5031 5032 buffer.frameCount = mFrameCount; 5033 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5034 size_t framesOut = buffer.frameCount; 5035 if (mResampler == NULL) { 5036 // no resampling 5037 while (framesOut) { 5038 size_t framesIn = mFrameCount - mRsmpInIndex; 5039 if (framesIn) { 5040 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5041 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5042 if (framesIn > framesOut) 5043 framesIn = framesOut; 5044 mRsmpInIndex += framesIn; 5045 framesOut -= framesIn; 5046 if ((int)mChannelCount == mReqChannelCount || 5047 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5048 memcpy(dst, src, framesIn * mFrameSize); 5049 } else { 5050 int16_t *src16 = (int16_t *)src; 5051 int16_t *dst16 = (int16_t *)dst; 5052 if (mChannelCount == 1) { 5053 while (framesIn--) { 5054 *dst16++ = *src16; 5055 *dst16++ = *src16++; 5056 } 5057 } else { 5058 while (framesIn--) { 5059 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5060 src16 += 2; 5061 } 5062 } 5063 } 5064 } 5065 if (framesOut && mFrameCount == mRsmpInIndex) { 5066 if (framesOut == mFrameCount && 5067 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5068 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5069 framesOut = 0; 5070 } else { 5071 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5072 mRsmpInIndex = 0; 5073 } 5074 if (mBytesRead < 0) { 5075 ALOGE("Error reading audio input"); 5076 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5077 // Force input into standby so that it tries to 5078 // recover at next read attempt 5079 mInput->stream->common.standby(&mInput->stream->common); 5080 usleep(kRecordThreadSleepUs); 5081 } 5082 mRsmpInIndex = mFrameCount; 5083 framesOut = 0; 5084 buffer.frameCount = 0; 5085 } 5086 } 5087 } 5088 } else { 5089 // resampling 5090 5091 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5092 // alter output frame count as if we were expecting stereo samples 5093 if (mChannelCount == 1 && mReqChannelCount == 1) { 5094 framesOut >>= 1; 5095 } 5096 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5097 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5098 // are 32 bit aligned which should be always true. 5099 if (mChannelCount == 2 && mReqChannelCount == 1) { 5100 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5101 // the resampler always outputs stereo samples: do post stereo to mono conversion 5102 int16_t *src = (int16_t *)mRsmpOutBuffer; 5103 int16_t *dst = buffer.i16; 5104 while (framesOut--) { 5105 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5106 src += 2; 5107 } 5108 } else { 5109 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5110 } 5111 5112 } 5113 if (mFramestoDrop == 0) { 5114 mActiveTrack->releaseBuffer(&buffer); 5115 } else { 5116 if (mFramestoDrop > 0) { 5117 mFramestoDrop -= buffer.frameCount; 5118 if (mFramestoDrop < 0) { 5119 mFramestoDrop = 0; 5120 } 5121 } 5122 } 5123 mActiveTrack->overflow(); 5124 } 5125 // client isn't retrieving buffers fast enough 5126 else { 5127 if (!mActiveTrack->setOverflow()) { 5128 nsecs_t now = systemTime(); 5129 if ((now - lastWarning) > kWarningThrottleNs) { 5130 ALOGW("RecordThread: buffer overflow"); 5131 lastWarning = now; 5132 } 5133 } 5134 // Release the processor for a while before asking for a new buffer. 5135 // This will give the application more chance to read from the buffer and 5136 // clear the overflow. 5137 usleep(kRecordThreadSleepUs); 5138 } 5139 } 5140 // enable changes in effect chain 5141 unlockEffectChains(effectChains); 5142 effectChains.clear(); 5143 } 5144 5145 if (!mStandby) { 5146 mInput->stream->common.standby(&mInput->stream->common); 5147 } 5148 mActiveTrack.clear(); 5149 5150 mStartStopCond.broadcast(); 5151 5152 releaseWakeLock(); 5153 5154 ALOGV("RecordThread %p exiting", this); 5155 return false; 5156} 5157 5158 5159sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5160 const sp<AudioFlinger::Client>& client, 5161 uint32_t sampleRate, 5162 audio_format_t format, 5163 int channelMask, 5164 int frameCount, 5165 int sessionId, 5166 status_t *status) 5167{ 5168 sp<RecordTrack> track; 5169 status_t lStatus; 5170 5171 lStatus = initCheck(); 5172 if (lStatus != NO_ERROR) { 5173 ALOGE("Audio driver not initialized."); 5174 goto Exit; 5175 } 5176 5177 { // scope for mLock 5178 Mutex::Autolock _l(mLock); 5179 5180 track = new RecordTrack(this, client, sampleRate, 5181 format, channelMask, frameCount, sessionId); 5182 5183 if (track->getCblk() == 0) { 5184 lStatus = NO_MEMORY; 5185 goto Exit; 5186 } 5187 5188 mTrack = track.get(); 5189 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5190 bool suspend = audio_is_bluetooth_sco_device( 5191 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5192 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5193 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5194 } 5195 lStatus = NO_ERROR; 5196 5197Exit: 5198 if (status) { 5199 *status = lStatus; 5200 } 5201 return track; 5202} 5203 5204status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5205 pid_t tid, AudioSystem::sync_event_t event, 5206 int triggerSession) 5207{ 5208 ALOGV("RecordThread::start tid=%d, event %d, triggerSession %d", tid, event, triggerSession); 5209 sp<ThreadBase> strongMe = this; 5210 status_t status = NO_ERROR; 5211 5212 if (event == AudioSystem::SYNC_EVENT_NONE) { 5213 mSyncStartEvent.clear(); 5214 mFramestoDrop = 0; 5215 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5216 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5217 triggerSession, 5218 recordTrack->sessionId(), 5219 syncStartEventCallback, 5220 this); 5221 mFramestoDrop = -1; 5222 } 5223 5224 { 5225 AutoMutex lock(mLock); 5226 if (mActiveTrack != 0) { 5227 if (recordTrack != mActiveTrack.get()) { 5228 status = -EBUSY; 5229 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5230 mActiveTrack->mState = TrackBase::ACTIVE; 5231 } 5232 return status; 5233 } 5234 5235 recordTrack->mState = TrackBase::IDLE; 5236 mActiveTrack = recordTrack; 5237 mLock.unlock(); 5238 status_t status = AudioSystem::startInput(mId); 5239 mLock.lock(); 5240 if (status != NO_ERROR) { 5241 mActiveTrack.clear(); 5242 clearSyncStartEvent(); 5243 return status; 5244 } 5245 mRsmpInIndex = mFrameCount; 5246 mBytesRead = 0; 5247 if (mResampler != NULL) { 5248 mResampler->reset(); 5249 } 5250 mActiveTrack->mState = TrackBase::RESUMING; 5251 // signal thread to start 5252 ALOGV("Signal record thread"); 5253 mWaitWorkCV.signal(); 5254 // do not wait for mStartStopCond if exiting 5255 if (exitPending()) { 5256 mActiveTrack.clear(); 5257 status = INVALID_OPERATION; 5258 goto startError; 5259 } 5260 mStartStopCond.wait(mLock); 5261 if (mActiveTrack == 0) { 5262 ALOGV("Record failed to start"); 5263 status = BAD_VALUE; 5264 goto startError; 5265 } 5266 ALOGV("Record started OK"); 5267 return status; 5268 } 5269startError: 5270 AudioSystem::stopInput(mId); 5271 clearSyncStartEvent(); 5272 return status; 5273} 5274 5275void AudioFlinger::RecordThread::clearSyncStartEvent() 5276{ 5277 if (mSyncStartEvent != 0) { 5278 mSyncStartEvent->cancel(); 5279 } 5280 mSyncStartEvent.clear(); 5281} 5282 5283void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5284{ 5285 sp<SyncEvent> strongEvent = event.promote(); 5286 5287 if (strongEvent != 0) { 5288 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5289 me->handleSyncStartEvent(strongEvent); 5290 } 5291} 5292 5293void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5294{ 5295 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5296 mActiveTrack.get(), 5297 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5298 event->listenerSession()); 5299 5300 if (mActiveTrack != 0 && 5301 event == mSyncStartEvent) { 5302 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5303 // from audio HAL 5304 mFramestoDrop = mFrameCount * 2; 5305 mSyncStartEvent.clear(); 5306 } 5307} 5308 5309void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5310 ALOGV("RecordThread::stop"); 5311 sp<ThreadBase> strongMe = this; 5312 { 5313 AutoMutex lock(mLock); 5314 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5315 mActiveTrack->mState = TrackBase::PAUSING; 5316 // do not wait for mStartStopCond if exiting 5317 if (exitPending()) { 5318 return; 5319 } 5320 mStartStopCond.wait(mLock); 5321 // if we have been restarted, recordTrack == mActiveTrack.get() here 5322 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5323 mLock.unlock(); 5324 AudioSystem::stopInput(mId); 5325 mLock.lock(); 5326 ALOGV("Record stopped OK"); 5327 } 5328 } 5329 } 5330} 5331 5332bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5333{ 5334 return false; 5335} 5336 5337status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5338{ 5339 if (!isValidSyncEvent(event)) { 5340 return BAD_VALUE; 5341 } 5342 5343 Mutex::Autolock _l(mLock); 5344 5345 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5346 mTrack->setSyncEvent(event); 5347 return NO_ERROR; 5348 } 5349 return NAME_NOT_FOUND; 5350} 5351 5352status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5353{ 5354 const size_t SIZE = 256; 5355 char buffer[SIZE]; 5356 String8 result; 5357 5358 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5359 result.append(buffer); 5360 5361 if (mActiveTrack != 0) { 5362 result.append("Active Track:\n"); 5363 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5364 mActiveTrack->dump(buffer, SIZE); 5365 result.append(buffer); 5366 5367 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5368 result.append(buffer); 5369 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5370 result.append(buffer); 5371 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5372 result.append(buffer); 5373 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5374 result.append(buffer); 5375 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5376 result.append(buffer); 5377 5378 5379 } else { 5380 result.append("No record client\n"); 5381 } 5382 write(fd, result.string(), result.size()); 5383 5384 dumpBase(fd, args); 5385 dumpEffectChains(fd, args); 5386 5387 return NO_ERROR; 5388} 5389 5390// AudioBufferProvider interface 5391status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5392{ 5393 size_t framesReq = buffer->frameCount; 5394 size_t framesReady = mFrameCount - mRsmpInIndex; 5395 int channelCount; 5396 5397 if (framesReady == 0) { 5398 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5399 if (mBytesRead < 0) { 5400 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5401 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5402 // Force input into standby so that it tries to 5403 // recover at next read attempt 5404 mInput->stream->common.standby(&mInput->stream->common); 5405 usleep(kRecordThreadSleepUs); 5406 } 5407 buffer->raw = NULL; 5408 buffer->frameCount = 0; 5409 return NOT_ENOUGH_DATA; 5410 } 5411 mRsmpInIndex = 0; 5412 framesReady = mFrameCount; 5413 } 5414 5415 if (framesReq > framesReady) { 5416 framesReq = framesReady; 5417 } 5418 5419 if (mChannelCount == 1 && mReqChannelCount == 2) { 5420 channelCount = 1; 5421 } else { 5422 channelCount = 2; 5423 } 5424 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5425 buffer->frameCount = framesReq; 5426 return NO_ERROR; 5427} 5428 5429// AudioBufferProvider interface 5430void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5431{ 5432 mRsmpInIndex += buffer->frameCount; 5433 buffer->frameCount = 0; 5434} 5435 5436bool AudioFlinger::RecordThread::checkForNewParameters_l() 5437{ 5438 bool reconfig = false; 5439 5440 while (!mNewParameters.isEmpty()) { 5441 status_t status = NO_ERROR; 5442 String8 keyValuePair = mNewParameters[0]; 5443 AudioParameter param = AudioParameter(keyValuePair); 5444 int value; 5445 audio_format_t reqFormat = mFormat; 5446 int reqSamplingRate = mReqSampleRate; 5447 int reqChannelCount = mReqChannelCount; 5448 5449 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5450 reqSamplingRate = value; 5451 reconfig = true; 5452 } 5453 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5454 reqFormat = (audio_format_t) value; 5455 reconfig = true; 5456 } 5457 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5458 reqChannelCount = popcount(value); 5459 reconfig = true; 5460 } 5461 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5462 // do not accept frame count changes if tracks are open as the track buffer 5463 // size depends on frame count and correct behavior would not be guaranteed 5464 // if frame count is changed after track creation 5465 if (mActiveTrack != 0) { 5466 status = INVALID_OPERATION; 5467 } else { 5468 reconfig = true; 5469 } 5470 } 5471 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5472 // forward device change to effects that have requested to be 5473 // aware of attached audio device. 5474 for (size_t i = 0; i < mEffectChains.size(); i++) { 5475 mEffectChains[i]->setDevice_l(value); 5476 } 5477 // store input device and output device but do not forward output device to audio HAL. 5478 // Note that status is ignored by the caller for output device 5479 // (see AudioFlinger::setParameters() 5480 if (value & AUDIO_DEVICE_OUT_ALL) { 5481 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5482 status = BAD_VALUE; 5483 } else { 5484 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5485 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5486 if (mTrack != NULL) { 5487 bool suspend = audio_is_bluetooth_sco_device( 5488 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5489 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5490 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5491 } 5492 } 5493 mDevice |= (uint32_t)value; 5494 } 5495 if (status == NO_ERROR) { 5496 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5497 if (status == INVALID_OPERATION) { 5498 mInput->stream->common.standby(&mInput->stream->common); 5499 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5500 keyValuePair.string()); 5501 } 5502 if (reconfig) { 5503 if (status == BAD_VALUE && 5504 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5505 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5506 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5507 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5508 (reqChannelCount <= FCC_2)) { 5509 status = NO_ERROR; 5510 } 5511 if (status == NO_ERROR) { 5512 readInputParameters(); 5513 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5514 } 5515 } 5516 } 5517 5518 mNewParameters.removeAt(0); 5519 5520 mParamStatus = status; 5521 mParamCond.signal(); 5522 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5523 // already timed out waiting for the status and will never signal the condition. 5524 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5525 } 5526 return reconfig; 5527} 5528 5529String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5530{ 5531 char *s; 5532 String8 out_s8 = String8(); 5533 5534 Mutex::Autolock _l(mLock); 5535 if (initCheck() != NO_ERROR) { 5536 return out_s8; 5537 } 5538 5539 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5540 out_s8 = String8(s); 5541 free(s); 5542 return out_s8; 5543} 5544 5545void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5546 AudioSystem::OutputDescriptor desc; 5547 void *param2 = NULL; 5548 5549 switch (event) { 5550 case AudioSystem::INPUT_OPENED: 5551 case AudioSystem::INPUT_CONFIG_CHANGED: 5552 desc.channels = mChannelMask; 5553 desc.samplingRate = mSampleRate; 5554 desc.format = mFormat; 5555 desc.frameCount = mFrameCount; 5556 desc.latency = 0; 5557 param2 = &desc; 5558 break; 5559 5560 case AudioSystem::INPUT_CLOSED: 5561 default: 5562 break; 5563 } 5564 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5565} 5566 5567void AudioFlinger::RecordThread::readInputParameters() 5568{ 5569 delete mRsmpInBuffer; 5570 // mRsmpInBuffer is always assigned a new[] below 5571 delete mRsmpOutBuffer; 5572 mRsmpOutBuffer = NULL; 5573 delete mResampler; 5574 mResampler = NULL; 5575 5576 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5577 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5578 mChannelCount = (uint16_t)popcount(mChannelMask); 5579 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5580 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5581 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5582 mFrameCount = mInputBytes / mFrameSize; 5583 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5584 5585 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5586 { 5587 int channelCount; 5588 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5589 // stereo to mono post process as the resampler always outputs stereo. 5590 if (mChannelCount == 1 && mReqChannelCount == 2) { 5591 channelCount = 1; 5592 } else { 5593 channelCount = 2; 5594 } 5595 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5596 mResampler->setSampleRate(mSampleRate); 5597 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5598 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5599 5600 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5601 if (mChannelCount == 1 && mReqChannelCount == 1) { 5602 mFrameCount >>= 1; 5603 } 5604 5605 } 5606 mRsmpInIndex = mFrameCount; 5607} 5608 5609unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5610{ 5611 Mutex::Autolock _l(mLock); 5612 if (initCheck() != NO_ERROR) { 5613 return 0; 5614 } 5615 5616 return mInput->stream->get_input_frames_lost(mInput->stream); 5617} 5618 5619uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5620{ 5621 Mutex::Autolock _l(mLock); 5622 uint32_t result = 0; 5623 if (getEffectChain_l(sessionId) != 0) { 5624 result = EFFECT_SESSION; 5625 } 5626 5627 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5628 result |= TRACK_SESSION; 5629 } 5630 5631 return result; 5632} 5633 5634AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5635{ 5636 Mutex::Autolock _l(mLock); 5637 return mTrack; 5638} 5639 5640AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5641{ 5642 Mutex::Autolock _l(mLock); 5643 return mInput; 5644} 5645 5646AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5647{ 5648 Mutex::Autolock _l(mLock); 5649 AudioStreamIn *input = mInput; 5650 mInput = NULL; 5651 return input; 5652} 5653 5654// this method must always be called either with ThreadBase mLock held or inside the thread loop 5655audio_stream_t* AudioFlinger::RecordThread::stream() const 5656{ 5657 if (mInput == NULL) { 5658 return NULL; 5659 } 5660 return &mInput->stream->common; 5661} 5662 5663 5664// ---------------------------------------------------------------------------- 5665 5666audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5667 uint32_t *pSamplingRate, 5668 audio_format_t *pFormat, 5669 uint32_t *pChannels, 5670 uint32_t *pLatencyMs, 5671 audio_policy_output_flags_t flags) 5672{ 5673 status_t status; 5674 PlaybackThread *thread = NULL; 5675 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5676 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5677 uint32_t channels = pChannels ? *pChannels : 0; 5678 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5679 audio_stream_out_t *outStream; 5680 audio_hw_device_t *outHwDev; 5681 5682 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5683 pDevices ? *pDevices : 0, 5684 samplingRate, 5685 format, 5686 channels, 5687 flags); 5688 5689 if (pDevices == NULL || *pDevices == 0) { 5690 return 0; 5691 } 5692 5693 Mutex::Autolock _l(mLock); 5694 5695 outHwDev = findSuitableHwDev_l(*pDevices); 5696 if (outHwDev == NULL) 5697 return 0; 5698 5699 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5700 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5701 &channels, &samplingRate, &outStream); 5702 mHardwareStatus = AUDIO_HW_IDLE; 5703 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5704 outStream, 5705 samplingRate, 5706 format, 5707 channels, 5708 status); 5709 5710 if (outStream != NULL) { 5711 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5712 audio_io_handle_t id = nextUniqueId(); 5713 5714 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5715 (format != AUDIO_FORMAT_PCM_16_BIT) || 5716 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5717 thread = new DirectOutputThread(this, output, id, *pDevices); 5718 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5719 } else { 5720 thread = new MixerThread(this, output, id, *pDevices); 5721 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5722 } 5723 mPlaybackThreads.add(id, thread); 5724 5725 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5726 if (pFormat != NULL) *pFormat = format; 5727 if (pChannels != NULL) *pChannels = channels; 5728 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5729 5730 // notify client processes of the new output creation 5731 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5732 return id; 5733 } 5734 5735 return 0; 5736} 5737 5738audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5739 audio_io_handle_t output2) 5740{ 5741 Mutex::Autolock _l(mLock); 5742 MixerThread *thread1 = checkMixerThread_l(output1); 5743 MixerThread *thread2 = checkMixerThread_l(output2); 5744 5745 if (thread1 == NULL || thread2 == NULL) { 5746 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5747 return 0; 5748 } 5749 5750 audio_io_handle_t id = nextUniqueId(); 5751 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5752 thread->addOutputTrack(thread2); 5753 mPlaybackThreads.add(id, thread); 5754 // notify client processes of the new output creation 5755 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5756 return id; 5757} 5758 5759status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5760{ 5761 // keep strong reference on the playback thread so that 5762 // it is not destroyed while exit() is executed 5763 sp<PlaybackThread> thread; 5764 { 5765 Mutex::Autolock _l(mLock); 5766 thread = checkPlaybackThread_l(output); 5767 if (thread == NULL) { 5768 return BAD_VALUE; 5769 } 5770 5771 ALOGV("closeOutput() %d", output); 5772 5773 if (thread->type() == ThreadBase::MIXER) { 5774 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5775 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5776 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5777 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5778 } 5779 } 5780 } 5781 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5782 mPlaybackThreads.removeItem(output); 5783 } 5784 thread->exit(); 5785 // The thread entity (active unit of execution) is no longer running here, 5786 // but the ThreadBase container still exists. 5787 5788 if (thread->type() != ThreadBase::DUPLICATING) { 5789 AudioStreamOut *out = thread->clearOutput(); 5790 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5791 // from now on thread->mOutput is NULL 5792 out->hwDev->close_output_stream(out->hwDev, out->stream); 5793 delete out; 5794 } 5795 return NO_ERROR; 5796} 5797 5798status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5799{ 5800 Mutex::Autolock _l(mLock); 5801 PlaybackThread *thread = checkPlaybackThread_l(output); 5802 5803 if (thread == NULL) { 5804 return BAD_VALUE; 5805 } 5806 5807 ALOGV("suspendOutput() %d", output); 5808 thread->suspend(); 5809 5810 return NO_ERROR; 5811} 5812 5813status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5814{ 5815 Mutex::Autolock _l(mLock); 5816 PlaybackThread *thread = checkPlaybackThread_l(output); 5817 5818 if (thread == NULL) { 5819 return BAD_VALUE; 5820 } 5821 5822 ALOGV("restoreOutput() %d", output); 5823 5824 thread->restore(); 5825 5826 return NO_ERROR; 5827} 5828 5829audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5830 uint32_t *pSamplingRate, 5831 audio_format_t *pFormat, 5832 uint32_t *pChannels, 5833 audio_in_acoustics_t acoustics) 5834{ 5835 status_t status; 5836 RecordThread *thread = NULL; 5837 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5838 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5839 uint32_t channels = pChannels ? *pChannels : 0; 5840 uint32_t reqSamplingRate = samplingRate; 5841 audio_format_t reqFormat = format; 5842 uint32_t reqChannels = channels; 5843 audio_stream_in_t *inStream; 5844 audio_hw_device_t *inHwDev; 5845 5846 if (pDevices == NULL || *pDevices == 0) { 5847 return 0; 5848 } 5849 5850 Mutex::Autolock _l(mLock); 5851 5852 inHwDev = findSuitableHwDev_l(*pDevices); 5853 if (inHwDev == NULL) 5854 return 0; 5855 5856 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5857 &channels, &samplingRate, 5858 acoustics, 5859 &inStream); 5860 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5861 inStream, 5862 samplingRate, 5863 format, 5864 channels, 5865 acoustics, 5866 status); 5867 5868 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5869 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5870 // or stereo to mono conversions on 16 bit PCM inputs. 5871 if (inStream == NULL && status == BAD_VALUE && 5872 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5873 (samplingRate <= 2 * reqSamplingRate) && 5874 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5875 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5876 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5877 &channels, &samplingRate, 5878 acoustics, 5879 &inStream); 5880 } 5881 5882 if (inStream != NULL) { 5883 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5884 5885 audio_io_handle_t id = nextUniqueId(); 5886 // Start record thread 5887 // RecorThread require both input and output device indication to forward to audio 5888 // pre processing modules 5889 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5890 thread = new RecordThread(this, 5891 input, 5892 reqSamplingRate, 5893 reqChannels, 5894 id, 5895 device); 5896 mRecordThreads.add(id, thread); 5897 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5898 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5899 if (pFormat != NULL) *pFormat = format; 5900 if (pChannels != NULL) *pChannels = reqChannels; 5901 5902 input->stream->common.standby(&input->stream->common); 5903 5904 // notify client processes of the new input creation 5905 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5906 return id; 5907 } 5908 5909 return 0; 5910} 5911 5912status_t AudioFlinger::closeInput(audio_io_handle_t input) 5913{ 5914 // keep strong reference on the record thread so that 5915 // it is not destroyed while exit() is executed 5916 sp<RecordThread> thread; 5917 { 5918 Mutex::Autolock _l(mLock); 5919 thread = checkRecordThread_l(input); 5920 if (thread == NULL) { 5921 return BAD_VALUE; 5922 } 5923 5924 ALOGV("closeInput() %d", input); 5925 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5926 mRecordThreads.removeItem(input); 5927 } 5928 thread->exit(); 5929 // The thread entity (active unit of execution) is no longer running here, 5930 // but the ThreadBase container still exists. 5931 5932 AudioStreamIn *in = thread->clearInput(); 5933 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5934 // from now on thread->mInput is NULL 5935 in->hwDev->close_input_stream(in->hwDev, in->stream); 5936 delete in; 5937 5938 return NO_ERROR; 5939} 5940 5941status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5942{ 5943 Mutex::Autolock _l(mLock); 5944 MixerThread *dstThread = checkMixerThread_l(output); 5945 if (dstThread == NULL) { 5946 ALOGW("setStreamOutput() bad output id %d", output); 5947 return BAD_VALUE; 5948 } 5949 5950 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5951 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5952 5953 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5954 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5955 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5956 MixerThread *srcThread = (MixerThread *)thread; 5957 srcThread->invalidateTracks(stream); 5958 } 5959 } 5960 5961 return NO_ERROR; 5962} 5963 5964 5965int AudioFlinger::newAudioSessionId() 5966{ 5967 return nextUniqueId(); 5968} 5969 5970void AudioFlinger::acquireAudioSessionId(int audioSession) 5971{ 5972 Mutex::Autolock _l(mLock); 5973 pid_t caller = IPCThreadState::self()->getCallingPid(); 5974 ALOGV("acquiring %d from %d", audioSession, caller); 5975 size_t num = mAudioSessionRefs.size(); 5976 for (size_t i = 0; i< num; i++) { 5977 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5978 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5979 ref->mCnt++; 5980 ALOGV(" incremented refcount to %d", ref->mCnt); 5981 return; 5982 } 5983 } 5984 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5985 ALOGV(" added new entry for %d", audioSession); 5986} 5987 5988void AudioFlinger::releaseAudioSessionId(int audioSession) 5989{ 5990 Mutex::Autolock _l(mLock); 5991 pid_t caller = IPCThreadState::self()->getCallingPid(); 5992 ALOGV("releasing %d from %d", audioSession, caller); 5993 size_t num = mAudioSessionRefs.size(); 5994 for (size_t i = 0; i< num; i++) { 5995 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5996 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5997 ref->mCnt--; 5998 ALOGV(" decremented refcount to %d", ref->mCnt); 5999 if (ref->mCnt == 0) { 6000 mAudioSessionRefs.removeAt(i); 6001 delete ref; 6002 purgeStaleEffects_l(); 6003 } 6004 return; 6005 } 6006 } 6007 ALOGW("session id %d not found for pid %d", audioSession, caller); 6008} 6009 6010void AudioFlinger::purgeStaleEffects_l() { 6011 6012 ALOGV("purging stale effects"); 6013 6014 Vector< sp<EffectChain> > chains; 6015 6016 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6017 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6018 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6019 sp<EffectChain> ec = t->mEffectChains[j]; 6020 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6021 chains.push(ec); 6022 } 6023 } 6024 } 6025 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6026 sp<RecordThread> t = mRecordThreads.valueAt(i); 6027 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6028 sp<EffectChain> ec = t->mEffectChains[j]; 6029 chains.push(ec); 6030 } 6031 } 6032 6033 for (size_t i = 0; i < chains.size(); i++) { 6034 sp<EffectChain> ec = chains[i]; 6035 int sessionid = ec->sessionId(); 6036 sp<ThreadBase> t = ec->mThread.promote(); 6037 if (t == 0) { 6038 continue; 6039 } 6040 size_t numsessionrefs = mAudioSessionRefs.size(); 6041 bool found = false; 6042 for (size_t k = 0; k < numsessionrefs; k++) { 6043 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6044 if (ref->mSessionid == sessionid) { 6045 ALOGV(" session %d still exists for %d with %d refs", 6046 sessionid, ref->mPid, ref->mCnt); 6047 found = true; 6048 break; 6049 } 6050 } 6051 if (!found) { 6052 // remove all effects from the chain 6053 while (ec->mEffects.size()) { 6054 sp<EffectModule> effect = ec->mEffects[0]; 6055 effect->unPin(); 6056 Mutex::Autolock _l (t->mLock); 6057 t->removeEffect_l(effect); 6058 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6059 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6060 if (handle != 0) { 6061 handle->mEffect.clear(); 6062 if (handle->mHasControl && handle->mEnabled) { 6063 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6064 } 6065 } 6066 } 6067 AudioSystem::unregisterEffect(effect->id()); 6068 } 6069 } 6070 } 6071 return; 6072} 6073 6074// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6075AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6076{ 6077 return mPlaybackThreads.valueFor(output).get(); 6078} 6079 6080// checkMixerThread_l() must be called with AudioFlinger::mLock held 6081AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6082{ 6083 PlaybackThread *thread = checkPlaybackThread_l(output); 6084 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6085} 6086 6087// checkRecordThread_l() must be called with AudioFlinger::mLock held 6088AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6089{ 6090 return mRecordThreads.valueFor(input).get(); 6091} 6092 6093uint32_t AudioFlinger::nextUniqueId() 6094{ 6095 return android_atomic_inc(&mNextUniqueId); 6096} 6097 6098AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6099{ 6100 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6101 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6102 AudioStreamOut *output = thread->getOutput(); 6103 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6104 return thread; 6105 } 6106 } 6107 return NULL; 6108} 6109 6110uint32_t AudioFlinger::primaryOutputDevice_l() const 6111{ 6112 PlaybackThread *thread = primaryPlaybackThread_l(); 6113 6114 if (thread == NULL) { 6115 return 0; 6116 } 6117 6118 return thread->device(); 6119} 6120 6121sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6122 int triggerSession, 6123 int listenerSession, 6124 sync_event_callback_t callBack, 6125 void *cookie) 6126{ 6127 Mutex::Autolock _l(mLock); 6128 6129 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6130 status_t playStatus = NAME_NOT_FOUND; 6131 status_t recStatus = NAME_NOT_FOUND; 6132 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6133 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6134 if (playStatus == NO_ERROR) { 6135 return event; 6136 } 6137 } 6138 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6139 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6140 if (recStatus == NO_ERROR) { 6141 return event; 6142 } 6143 } 6144 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6145 mPendingSyncEvents.add(event); 6146 } else { 6147 ALOGV("createSyncEvent() invalid event %d", event->type()); 6148 event.clear(); 6149 } 6150 return event; 6151} 6152 6153// ---------------------------------------------------------------------------- 6154// Effect management 6155// ---------------------------------------------------------------------------- 6156 6157 6158status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6159{ 6160 Mutex::Autolock _l(mLock); 6161 return EffectQueryNumberEffects(numEffects); 6162} 6163 6164status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6165{ 6166 Mutex::Autolock _l(mLock); 6167 return EffectQueryEffect(index, descriptor); 6168} 6169 6170status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6171 effect_descriptor_t *descriptor) const 6172{ 6173 Mutex::Autolock _l(mLock); 6174 return EffectGetDescriptor(pUuid, descriptor); 6175} 6176 6177 6178sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6179 effect_descriptor_t *pDesc, 6180 const sp<IEffectClient>& effectClient, 6181 int32_t priority, 6182 audio_io_handle_t io, 6183 int sessionId, 6184 status_t *status, 6185 int *id, 6186 int *enabled) 6187{ 6188 status_t lStatus = NO_ERROR; 6189 sp<EffectHandle> handle; 6190 effect_descriptor_t desc; 6191 6192 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6193 pid, effectClient.get(), priority, sessionId, io); 6194 6195 if (pDesc == NULL) { 6196 lStatus = BAD_VALUE; 6197 goto Exit; 6198 } 6199 6200 // check audio settings permission for global effects 6201 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6202 lStatus = PERMISSION_DENIED; 6203 goto Exit; 6204 } 6205 6206 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6207 // that can only be created by audio policy manager (running in same process) 6208 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6209 lStatus = PERMISSION_DENIED; 6210 goto Exit; 6211 } 6212 6213 if (io == 0) { 6214 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6215 // output must be specified by AudioPolicyManager when using session 6216 // AUDIO_SESSION_OUTPUT_STAGE 6217 lStatus = BAD_VALUE; 6218 goto Exit; 6219 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6220 // if the output returned by getOutputForEffect() is removed before we lock the 6221 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6222 // and we will exit safely 6223 io = AudioSystem::getOutputForEffect(&desc); 6224 } 6225 } 6226 6227 { 6228 Mutex::Autolock _l(mLock); 6229 6230 6231 if (!EffectIsNullUuid(&pDesc->uuid)) { 6232 // if uuid is specified, request effect descriptor 6233 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6234 if (lStatus < 0) { 6235 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6236 goto Exit; 6237 } 6238 } else { 6239 // if uuid is not specified, look for an available implementation 6240 // of the required type in effect factory 6241 if (EffectIsNullUuid(&pDesc->type)) { 6242 ALOGW("createEffect() no effect type"); 6243 lStatus = BAD_VALUE; 6244 goto Exit; 6245 } 6246 uint32_t numEffects = 0; 6247 effect_descriptor_t d; 6248 d.flags = 0; // prevent compiler warning 6249 bool found = false; 6250 6251 lStatus = EffectQueryNumberEffects(&numEffects); 6252 if (lStatus < 0) { 6253 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6254 goto Exit; 6255 } 6256 for (uint32_t i = 0; i < numEffects; i++) { 6257 lStatus = EffectQueryEffect(i, &desc); 6258 if (lStatus < 0) { 6259 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6260 continue; 6261 } 6262 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6263 // If matching type found save effect descriptor. If the session is 6264 // 0 and the effect is not auxiliary, continue enumeration in case 6265 // an auxiliary version of this effect type is available 6266 found = true; 6267 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6268 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6269 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6270 break; 6271 } 6272 } 6273 } 6274 if (!found) { 6275 lStatus = BAD_VALUE; 6276 ALOGW("createEffect() effect not found"); 6277 goto Exit; 6278 } 6279 // For same effect type, chose auxiliary version over insert version if 6280 // connect to output mix (Compliance to OpenSL ES) 6281 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6282 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6283 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6284 } 6285 } 6286 6287 // Do not allow auxiliary effects on a session different from 0 (output mix) 6288 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6289 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6290 lStatus = INVALID_OPERATION; 6291 goto Exit; 6292 } 6293 6294 // check recording permission for visualizer 6295 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6296 !recordingAllowed()) { 6297 lStatus = PERMISSION_DENIED; 6298 goto Exit; 6299 } 6300 6301 // return effect descriptor 6302 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6303 6304 // If output is not specified try to find a matching audio session ID in one of the 6305 // output threads. 6306 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6307 // because of code checking output when entering the function. 6308 // Note: io is never 0 when creating an effect on an input 6309 if (io == 0) { 6310 // look for the thread where the specified audio session is present 6311 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6312 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6313 io = mPlaybackThreads.keyAt(i); 6314 break; 6315 } 6316 } 6317 if (io == 0) { 6318 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6319 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6320 io = mRecordThreads.keyAt(i); 6321 break; 6322 } 6323 } 6324 } 6325 // If no output thread contains the requested session ID, default to 6326 // first output. The effect chain will be moved to the correct output 6327 // thread when a track with the same session ID is created 6328 if (io == 0 && mPlaybackThreads.size()) { 6329 io = mPlaybackThreads.keyAt(0); 6330 } 6331 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6332 } 6333 ThreadBase *thread = checkRecordThread_l(io); 6334 if (thread == NULL) { 6335 thread = checkPlaybackThread_l(io); 6336 if (thread == NULL) { 6337 ALOGE("createEffect() unknown output thread"); 6338 lStatus = BAD_VALUE; 6339 goto Exit; 6340 } 6341 } 6342 6343 sp<Client> client = registerPid_l(pid); 6344 6345 // create effect on selected output thread 6346 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6347 &desc, enabled, &lStatus); 6348 if (handle != 0 && id != NULL) { 6349 *id = handle->id(); 6350 } 6351 } 6352 6353Exit: 6354 if (status != NULL) { 6355 *status = lStatus; 6356 } 6357 return handle; 6358} 6359 6360status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6361 audio_io_handle_t dstOutput) 6362{ 6363 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6364 sessionId, srcOutput, dstOutput); 6365 Mutex::Autolock _l(mLock); 6366 if (srcOutput == dstOutput) { 6367 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6368 return NO_ERROR; 6369 } 6370 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6371 if (srcThread == NULL) { 6372 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6373 return BAD_VALUE; 6374 } 6375 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6376 if (dstThread == NULL) { 6377 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6378 return BAD_VALUE; 6379 } 6380 6381 Mutex::Autolock _dl(dstThread->mLock); 6382 Mutex::Autolock _sl(srcThread->mLock); 6383 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6384 6385 return NO_ERROR; 6386} 6387 6388// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6389status_t AudioFlinger::moveEffectChain_l(int sessionId, 6390 AudioFlinger::PlaybackThread *srcThread, 6391 AudioFlinger::PlaybackThread *dstThread, 6392 bool reRegister) 6393{ 6394 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6395 sessionId, srcThread, dstThread); 6396 6397 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6398 if (chain == 0) { 6399 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6400 sessionId, srcThread); 6401 return INVALID_OPERATION; 6402 } 6403 6404 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6405 // so that a new chain is created with correct parameters when first effect is added. This is 6406 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6407 // removed. 6408 srcThread->removeEffectChain_l(chain); 6409 6410 // transfer all effects one by one so that new effect chain is created on new thread with 6411 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6412 audio_io_handle_t dstOutput = dstThread->id(); 6413 sp<EffectChain> dstChain; 6414 uint32_t strategy = 0; // prevent compiler warning 6415 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6416 while (effect != 0) { 6417 srcThread->removeEffect_l(effect); 6418 dstThread->addEffect_l(effect); 6419 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6420 if (effect->state() == EffectModule::ACTIVE || 6421 effect->state() == EffectModule::STOPPING) { 6422 effect->start(); 6423 } 6424 // if the move request is not received from audio policy manager, the effect must be 6425 // re-registered with the new strategy and output 6426 if (dstChain == 0) { 6427 dstChain = effect->chain().promote(); 6428 if (dstChain == 0) { 6429 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6430 srcThread->addEffect_l(effect); 6431 return NO_INIT; 6432 } 6433 strategy = dstChain->strategy(); 6434 } 6435 if (reRegister) { 6436 AudioSystem::unregisterEffect(effect->id()); 6437 AudioSystem::registerEffect(&effect->desc(), 6438 dstOutput, 6439 strategy, 6440 sessionId, 6441 effect->id()); 6442 } 6443 effect = chain->getEffectFromId_l(0); 6444 } 6445 6446 return NO_ERROR; 6447} 6448 6449 6450// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6451sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6452 const sp<AudioFlinger::Client>& client, 6453 const sp<IEffectClient>& effectClient, 6454 int32_t priority, 6455 int sessionId, 6456 effect_descriptor_t *desc, 6457 int *enabled, 6458 status_t *status 6459 ) 6460{ 6461 sp<EffectModule> effect; 6462 sp<EffectHandle> handle; 6463 status_t lStatus; 6464 sp<EffectChain> chain; 6465 bool chainCreated = false; 6466 bool effectCreated = false; 6467 bool effectRegistered = false; 6468 6469 lStatus = initCheck(); 6470 if (lStatus != NO_ERROR) { 6471 ALOGW("createEffect_l() Audio driver not initialized."); 6472 goto Exit; 6473 } 6474 6475 // Do not allow effects with session ID 0 on direct output or duplicating threads 6476 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6477 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6478 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6479 desc->name, sessionId); 6480 lStatus = BAD_VALUE; 6481 goto Exit; 6482 } 6483 // Only Pre processor effects are allowed on input threads and only on input threads 6484 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6485 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6486 desc->name, desc->flags, mType); 6487 lStatus = BAD_VALUE; 6488 goto Exit; 6489 } 6490 6491 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6492 6493 { // scope for mLock 6494 Mutex::Autolock _l(mLock); 6495 6496 // check for existing effect chain with the requested audio session 6497 chain = getEffectChain_l(sessionId); 6498 if (chain == 0) { 6499 // create a new chain for this session 6500 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6501 chain = new EffectChain(this, sessionId); 6502 addEffectChain_l(chain); 6503 chain->setStrategy(getStrategyForSession_l(sessionId)); 6504 chainCreated = true; 6505 } else { 6506 effect = chain->getEffectFromDesc_l(desc); 6507 } 6508 6509 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6510 6511 if (effect == 0) { 6512 int id = mAudioFlinger->nextUniqueId(); 6513 // Check CPU and memory usage 6514 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6515 if (lStatus != NO_ERROR) { 6516 goto Exit; 6517 } 6518 effectRegistered = true; 6519 // create a new effect module if none present in the chain 6520 effect = new EffectModule(this, chain, desc, id, sessionId); 6521 lStatus = effect->status(); 6522 if (lStatus != NO_ERROR) { 6523 goto Exit; 6524 } 6525 lStatus = chain->addEffect_l(effect); 6526 if (lStatus != NO_ERROR) { 6527 goto Exit; 6528 } 6529 effectCreated = true; 6530 6531 effect->setDevice(mDevice); 6532 effect->setMode(mAudioFlinger->getMode()); 6533 } 6534 // create effect handle and connect it to effect module 6535 handle = new EffectHandle(effect, client, effectClient, priority); 6536 lStatus = effect->addHandle(handle); 6537 if (enabled != NULL) { 6538 *enabled = (int)effect->isEnabled(); 6539 } 6540 } 6541 6542Exit: 6543 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6544 Mutex::Autolock _l(mLock); 6545 if (effectCreated) { 6546 chain->removeEffect_l(effect); 6547 } 6548 if (effectRegistered) { 6549 AudioSystem::unregisterEffect(effect->id()); 6550 } 6551 if (chainCreated) { 6552 removeEffectChain_l(chain); 6553 } 6554 handle.clear(); 6555 } 6556 6557 if (status != NULL) { 6558 *status = lStatus; 6559 } 6560 return handle; 6561} 6562 6563sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6564{ 6565 sp<EffectChain> chain = getEffectChain_l(sessionId); 6566 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6567} 6568 6569// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6570// PlaybackThread::mLock held 6571status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6572{ 6573 // check for existing effect chain with the requested audio session 6574 int sessionId = effect->sessionId(); 6575 sp<EffectChain> chain = getEffectChain_l(sessionId); 6576 bool chainCreated = false; 6577 6578 if (chain == 0) { 6579 // create a new chain for this session 6580 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6581 chain = new EffectChain(this, sessionId); 6582 addEffectChain_l(chain); 6583 chain->setStrategy(getStrategyForSession_l(sessionId)); 6584 chainCreated = true; 6585 } 6586 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6587 6588 if (chain->getEffectFromId_l(effect->id()) != 0) { 6589 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6590 this, effect->desc().name, chain.get()); 6591 return BAD_VALUE; 6592 } 6593 6594 status_t status = chain->addEffect_l(effect); 6595 if (status != NO_ERROR) { 6596 if (chainCreated) { 6597 removeEffectChain_l(chain); 6598 } 6599 return status; 6600 } 6601 6602 effect->setDevice(mDevice); 6603 effect->setMode(mAudioFlinger->getMode()); 6604 return NO_ERROR; 6605} 6606 6607void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6608 6609 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6610 effect_descriptor_t desc = effect->desc(); 6611 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6612 detachAuxEffect_l(effect->id()); 6613 } 6614 6615 sp<EffectChain> chain = effect->chain().promote(); 6616 if (chain != 0) { 6617 // remove effect chain if removing last effect 6618 if (chain->removeEffect_l(effect) == 0) { 6619 removeEffectChain_l(chain); 6620 } 6621 } else { 6622 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6623 } 6624} 6625 6626void AudioFlinger::ThreadBase::lockEffectChains_l( 6627 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6628{ 6629 effectChains = mEffectChains; 6630 for (size_t i = 0; i < mEffectChains.size(); i++) { 6631 mEffectChains[i]->lock(); 6632 } 6633} 6634 6635void AudioFlinger::ThreadBase::unlockEffectChains( 6636 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6637{ 6638 for (size_t i = 0; i < effectChains.size(); i++) { 6639 effectChains[i]->unlock(); 6640 } 6641} 6642 6643sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6644{ 6645 Mutex::Autolock _l(mLock); 6646 return getEffectChain_l(sessionId); 6647} 6648 6649sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6650{ 6651 size_t size = mEffectChains.size(); 6652 for (size_t i = 0; i < size; i++) { 6653 if (mEffectChains[i]->sessionId() == sessionId) { 6654 return mEffectChains[i]; 6655 } 6656 } 6657 return 0; 6658} 6659 6660void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6661{ 6662 Mutex::Autolock _l(mLock); 6663 size_t size = mEffectChains.size(); 6664 for (size_t i = 0; i < size; i++) { 6665 mEffectChains[i]->setMode_l(mode); 6666 } 6667} 6668 6669void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6670 const wp<EffectHandle>& handle, 6671 bool unpinIfLast) { 6672 6673 Mutex::Autolock _l(mLock); 6674 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6675 // delete the effect module if removing last handle on it 6676 if (effect->removeHandle(handle) == 0) { 6677 if (!effect->isPinned() || unpinIfLast) { 6678 removeEffect_l(effect); 6679 AudioSystem::unregisterEffect(effect->id()); 6680 } 6681 } 6682} 6683 6684status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6685{ 6686 int session = chain->sessionId(); 6687 int16_t *buffer = mMixBuffer; 6688 bool ownsBuffer = false; 6689 6690 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6691 if (session > 0) { 6692 // Only one effect chain can be present in direct output thread and it uses 6693 // the mix buffer as input 6694 if (mType != DIRECT) { 6695 size_t numSamples = mFrameCount * mChannelCount; 6696 buffer = new int16_t[numSamples]; 6697 memset(buffer, 0, numSamples * sizeof(int16_t)); 6698 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6699 ownsBuffer = true; 6700 } 6701 6702 // Attach all tracks with same session ID to this chain. 6703 for (size_t i = 0; i < mTracks.size(); ++i) { 6704 sp<Track> track = mTracks[i]; 6705 if (session == track->sessionId()) { 6706 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6707 track->setMainBuffer(buffer); 6708 chain->incTrackCnt(); 6709 } 6710 } 6711 6712 // indicate all active tracks in the chain 6713 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6714 sp<Track> track = mActiveTracks[i].promote(); 6715 if (track == 0) continue; 6716 if (session == track->sessionId()) { 6717 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6718 chain->incActiveTrackCnt(); 6719 } 6720 } 6721 } 6722 6723 chain->setInBuffer(buffer, ownsBuffer); 6724 chain->setOutBuffer(mMixBuffer); 6725 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6726 // chains list in order to be processed last as it contains output stage effects 6727 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6728 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6729 // after track specific effects and before output stage 6730 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6731 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6732 // Effect chain for other sessions are inserted at beginning of effect 6733 // chains list to be processed before output mix effects. Relative order between other 6734 // sessions is not important 6735 size_t size = mEffectChains.size(); 6736 size_t i = 0; 6737 for (i = 0; i < size; i++) { 6738 if (mEffectChains[i]->sessionId() < session) break; 6739 } 6740 mEffectChains.insertAt(chain, i); 6741 checkSuspendOnAddEffectChain_l(chain); 6742 6743 return NO_ERROR; 6744} 6745 6746size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6747{ 6748 int session = chain->sessionId(); 6749 6750 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6751 6752 for (size_t i = 0; i < mEffectChains.size(); i++) { 6753 if (chain == mEffectChains[i]) { 6754 mEffectChains.removeAt(i); 6755 // detach all active tracks from the chain 6756 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6757 sp<Track> track = mActiveTracks[i].promote(); 6758 if (track == 0) continue; 6759 if (session == track->sessionId()) { 6760 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6761 chain.get(), session); 6762 chain->decActiveTrackCnt(); 6763 } 6764 } 6765 6766 // detach all tracks with same session ID from this chain 6767 for (size_t i = 0; i < mTracks.size(); ++i) { 6768 sp<Track> track = mTracks[i]; 6769 if (session == track->sessionId()) { 6770 track->setMainBuffer(mMixBuffer); 6771 chain->decTrackCnt(); 6772 } 6773 } 6774 break; 6775 } 6776 } 6777 return mEffectChains.size(); 6778} 6779 6780status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6781 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6782{ 6783 Mutex::Autolock _l(mLock); 6784 return attachAuxEffect_l(track, EffectId); 6785} 6786 6787status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6788 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6789{ 6790 status_t status = NO_ERROR; 6791 6792 if (EffectId == 0) { 6793 track->setAuxBuffer(0, NULL); 6794 } else { 6795 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6796 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6797 if (effect != 0) { 6798 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6799 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6800 } else { 6801 status = INVALID_OPERATION; 6802 } 6803 } else { 6804 status = BAD_VALUE; 6805 } 6806 } 6807 return status; 6808} 6809 6810void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6811{ 6812 for (size_t i = 0; i < mTracks.size(); ++i) { 6813 sp<Track> track = mTracks[i]; 6814 if (track->auxEffectId() == effectId) { 6815 attachAuxEffect_l(track, 0); 6816 } 6817 } 6818} 6819 6820status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6821{ 6822 // only one chain per input thread 6823 if (mEffectChains.size() != 0) { 6824 return INVALID_OPERATION; 6825 } 6826 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6827 6828 chain->setInBuffer(NULL); 6829 chain->setOutBuffer(NULL); 6830 6831 checkSuspendOnAddEffectChain_l(chain); 6832 6833 mEffectChains.add(chain); 6834 6835 return NO_ERROR; 6836} 6837 6838size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6839{ 6840 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6841 ALOGW_IF(mEffectChains.size() != 1, 6842 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6843 chain.get(), mEffectChains.size(), this); 6844 if (mEffectChains.size() == 1) { 6845 mEffectChains.removeAt(0); 6846 } 6847 return 0; 6848} 6849 6850// ---------------------------------------------------------------------------- 6851// EffectModule implementation 6852// ---------------------------------------------------------------------------- 6853 6854#undef LOG_TAG 6855#define LOG_TAG "AudioFlinger::EffectModule" 6856 6857AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6858 const wp<AudioFlinger::EffectChain>& chain, 6859 effect_descriptor_t *desc, 6860 int id, 6861 int sessionId) 6862 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6863 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6864{ 6865 ALOGV("Constructor %p", this); 6866 int lStatus; 6867 if (thread == NULL) { 6868 return; 6869 } 6870 6871 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6872 6873 // create effect engine from effect factory 6874 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6875 6876 if (mStatus != NO_ERROR) { 6877 return; 6878 } 6879 lStatus = init(); 6880 if (lStatus < 0) { 6881 mStatus = lStatus; 6882 goto Error; 6883 } 6884 6885 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6886 mPinned = true; 6887 } 6888 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6889 return; 6890Error: 6891 EffectRelease(mEffectInterface); 6892 mEffectInterface = NULL; 6893 ALOGV("Constructor Error %d", mStatus); 6894} 6895 6896AudioFlinger::EffectModule::~EffectModule() 6897{ 6898 ALOGV("Destructor %p", this); 6899 if (mEffectInterface != NULL) { 6900 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6901 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6902 sp<ThreadBase> thread = mThread.promote(); 6903 if (thread != 0) { 6904 audio_stream_t *stream = thread->stream(); 6905 if (stream != NULL) { 6906 stream->remove_audio_effect(stream, mEffectInterface); 6907 } 6908 } 6909 } 6910 // release effect engine 6911 EffectRelease(mEffectInterface); 6912 } 6913} 6914 6915status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6916{ 6917 status_t status; 6918 6919 Mutex::Autolock _l(mLock); 6920 int priority = handle->priority(); 6921 size_t size = mHandles.size(); 6922 sp<EffectHandle> h; 6923 size_t i; 6924 for (i = 0; i < size; i++) { 6925 h = mHandles[i].promote(); 6926 if (h == 0) continue; 6927 if (h->priority() <= priority) break; 6928 } 6929 // if inserted in first place, move effect control from previous owner to this handle 6930 if (i == 0) { 6931 bool enabled = false; 6932 if (h != 0) { 6933 enabled = h->enabled(); 6934 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6935 } 6936 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6937 status = NO_ERROR; 6938 } else { 6939 status = ALREADY_EXISTS; 6940 } 6941 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6942 mHandles.insertAt(handle, i); 6943 return status; 6944} 6945 6946size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6947{ 6948 Mutex::Autolock _l(mLock); 6949 size_t size = mHandles.size(); 6950 size_t i; 6951 for (i = 0; i < size; i++) { 6952 if (mHandles[i] == handle) break; 6953 } 6954 if (i == size) { 6955 return size; 6956 } 6957 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6958 6959 bool enabled = false; 6960 EffectHandle *hdl = handle.unsafe_get(); 6961 if (hdl != NULL) { 6962 ALOGV("removeHandle() unsafe_get OK"); 6963 enabled = hdl->enabled(); 6964 } 6965 mHandles.removeAt(i); 6966 size = mHandles.size(); 6967 // if removed from first place, move effect control from this handle to next in line 6968 if (i == 0 && size != 0) { 6969 sp<EffectHandle> h = mHandles[0].promote(); 6970 if (h != 0) { 6971 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6972 } 6973 } 6974 6975 // Prevent calls to process() and other functions on effect interface from now on. 6976 // The effect engine will be released by the destructor when the last strong reference on 6977 // this object is released which can happen after next process is called. 6978 if (size == 0 && !mPinned) { 6979 mState = DESTROYED; 6980 } 6981 6982 return size; 6983} 6984 6985sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6986{ 6987 Mutex::Autolock _l(mLock); 6988 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6989} 6990 6991void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6992{ 6993 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6994 // keep a strong reference on this EffectModule to avoid calling the 6995 // destructor before we exit 6996 sp<EffectModule> keep(this); 6997 { 6998 sp<ThreadBase> thread = mThread.promote(); 6999 if (thread != 0) { 7000 thread->disconnectEffect(keep, handle, unpinIfLast); 7001 } 7002 } 7003} 7004 7005void AudioFlinger::EffectModule::updateState() { 7006 Mutex::Autolock _l(mLock); 7007 7008 switch (mState) { 7009 case RESTART: 7010 reset_l(); 7011 // FALL THROUGH 7012 7013 case STARTING: 7014 // clear auxiliary effect input buffer for next accumulation 7015 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7016 memset(mConfig.inputCfg.buffer.raw, 7017 0, 7018 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7019 } 7020 start_l(); 7021 mState = ACTIVE; 7022 break; 7023 case STOPPING: 7024 stop_l(); 7025 mDisableWaitCnt = mMaxDisableWaitCnt; 7026 mState = STOPPED; 7027 break; 7028 case STOPPED: 7029 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7030 // turn off sequence. 7031 if (--mDisableWaitCnt == 0) { 7032 reset_l(); 7033 mState = IDLE; 7034 } 7035 break; 7036 default: //IDLE , ACTIVE, DESTROYED 7037 break; 7038 } 7039} 7040 7041void AudioFlinger::EffectModule::process() 7042{ 7043 Mutex::Autolock _l(mLock); 7044 7045 if (mState == DESTROYED || mEffectInterface == NULL || 7046 mConfig.inputCfg.buffer.raw == NULL || 7047 mConfig.outputCfg.buffer.raw == NULL) { 7048 return; 7049 } 7050 7051 if (isProcessEnabled()) { 7052 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7053 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7054 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7055 mConfig.inputCfg.buffer.s32, 7056 mConfig.inputCfg.buffer.frameCount/2); 7057 } 7058 7059 // do the actual processing in the effect engine 7060 int ret = (*mEffectInterface)->process(mEffectInterface, 7061 &mConfig.inputCfg.buffer, 7062 &mConfig.outputCfg.buffer); 7063 7064 // force transition to IDLE state when engine is ready 7065 if (mState == STOPPED && ret == -ENODATA) { 7066 mDisableWaitCnt = 1; 7067 } 7068 7069 // clear auxiliary effect input buffer for next accumulation 7070 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7071 memset(mConfig.inputCfg.buffer.raw, 0, 7072 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7073 } 7074 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7075 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7076 // If an insert effect is idle and input buffer is different from output buffer, 7077 // accumulate input onto output 7078 sp<EffectChain> chain = mChain.promote(); 7079 if (chain != 0 && chain->activeTrackCnt() != 0) { 7080 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7081 int16_t *in = mConfig.inputCfg.buffer.s16; 7082 int16_t *out = mConfig.outputCfg.buffer.s16; 7083 for (size_t i = 0; i < frameCnt; i++) { 7084 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7085 } 7086 } 7087 } 7088} 7089 7090void AudioFlinger::EffectModule::reset_l() 7091{ 7092 if (mEffectInterface == NULL) { 7093 return; 7094 } 7095 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7096} 7097 7098status_t AudioFlinger::EffectModule::configure() 7099{ 7100 uint32_t channels; 7101 if (mEffectInterface == NULL) { 7102 return NO_INIT; 7103 } 7104 7105 sp<ThreadBase> thread = mThread.promote(); 7106 if (thread == 0) { 7107 return DEAD_OBJECT; 7108 } 7109 7110 // TODO: handle configuration of effects replacing track process 7111 if (thread->channelCount() == 1) { 7112 channels = AUDIO_CHANNEL_OUT_MONO; 7113 } else { 7114 channels = AUDIO_CHANNEL_OUT_STEREO; 7115 } 7116 7117 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7118 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7119 } else { 7120 mConfig.inputCfg.channels = channels; 7121 } 7122 mConfig.outputCfg.channels = channels; 7123 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7124 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7125 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7126 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7127 mConfig.inputCfg.bufferProvider.cookie = NULL; 7128 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7129 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7130 mConfig.outputCfg.bufferProvider.cookie = NULL; 7131 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7132 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7133 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7134 // Insert effect: 7135 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7136 // always overwrites output buffer: input buffer == output buffer 7137 // - in other sessions: 7138 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7139 // other effect: overwrites output buffer: input buffer == output buffer 7140 // Auxiliary effect: 7141 // accumulates in output buffer: input buffer != output buffer 7142 // Therefore: accumulate <=> input buffer != output buffer 7143 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7144 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7145 } else { 7146 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7147 } 7148 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7149 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7150 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7151 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7152 7153 ALOGV("configure() %p thread %p buffer %p framecount %d", 7154 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7155 7156 status_t cmdStatus; 7157 uint32_t size = sizeof(int); 7158 status_t status = (*mEffectInterface)->command(mEffectInterface, 7159 EFFECT_CMD_SET_CONFIG, 7160 sizeof(effect_config_t), 7161 &mConfig, 7162 &size, 7163 &cmdStatus); 7164 if (status == 0) { 7165 status = cmdStatus; 7166 } 7167 7168 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7169 (1000 * mConfig.outputCfg.buffer.frameCount); 7170 7171 return status; 7172} 7173 7174status_t AudioFlinger::EffectModule::init() 7175{ 7176 Mutex::Autolock _l(mLock); 7177 if (mEffectInterface == NULL) { 7178 return NO_INIT; 7179 } 7180 status_t cmdStatus; 7181 uint32_t size = sizeof(status_t); 7182 status_t status = (*mEffectInterface)->command(mEffectInterface, 7183 EFFECT_CMD_INIT, 7184 0, 7185 NULL, 7186 &size, 7187 &cmdStatus); 7188 if (status == 0) { 7189 status = cmdStatus; 7190 } 7191 return status; 7192} 7193 7194status_t AudioFlinger::EffectModule::start() 7195{ 7196 Mutex::Autolock _l(mLock); 7197 return start_l(); 7198} 7199 7200status_t AudioFlinger::EffectModule::start_l() 7201{ 7202 if (mEffectInterface == NULL) { 7203 return NO_INIT; 7204 } 7205 status_t cmdStatus; 7206 uint32_t size = sizeof(status_t); 7207 status_t status = (*mEffectInterface)->command(mEffectInterface, 7208 EFFECT_CMD_ENABLE, 7209 0, 7210 NULL, 7211 &size, 7212 &cmdStatus); 7213 if (status == 0) { 7214 status = cmdStatus; 7215 } 7216 if (status == 0 && 7217 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7218 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7219 sp<ThreadBase> thread = mThread.promote(); 7220 if (thread != 0) { 7221 audio_stream_t *stream = thread->stream(); 7222 if (stream != NULL) { 7223 stream->add_audio_effect(stream, mEffectInterface); 7224 } 7225 } 7226 } 7227 return status; 7228} 7229 7230status_t AudioFlinger::EffectModule::stop() 7231{ 7232 Mutex::Autolock _l(mLock); 7233 return stop_l(); 7234} 7235 7236status_t AudioFlinger::EffectModule::stop_l() 7237{ 7238 if (mEffectInterface == NULL) { 7239 return NO_INIT; 7240 } 7241 status_t cmdStatus; 7242 uint32_t size = sizeof(status_t); 7243 status_t status = (*mEffectInterface)->command(mEffectInterface, 7244 EFFECT_CMD_DISABLE, 7245 0, 7246 NULL, 7247 &size, 7248 &cmdStatus); 7249 if (status == 0) { 7250 status = cmdStatus; 7251 } 7252 if (status == 0 && 7253 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7254 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7255 sp<ThreadBase> thread = mThread.promote(); 7256 if (thread != 0) { 7257 audio_stream_t *stream = thread->stream(); 7258 if (stream != NULL) { 7259 stream->remove_audio_effect(stream, mEffectInterface); 7260 } 7261 } 7262 } 7263 return status; 7264} 7265 7266status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7267 uint32_t cmdSize, 7268 void *pCmdData, 7269 uint32_t *replySize, 7270 void *pReplyData) 7271{ 7272 Mutex::Autolock _l(mLock); 7273// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7274 7275 if (mState == DESTROYED || mEffectInterface == NULL) { 7276 return NO_INIT; 7277 } 7278 status_t status = (*mEffectInterface)->command(mEffectInterface, 7279 cmdCode, 7280 cmdSize, 7281 pCmdData, 7282 replySize, 7283 pReplyData); 7284 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7285 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7286 for (size_t i = 1; i < mHandles.size(); i++) { 7287 sp<EffectHandle> h = mHandles[i].promote(); 7288 if (h != 0) { 7289 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7290 } 7291 } 7292 } 7293 return status; 7294} 7295 7296status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7297{ 7298 7299 Mutex::Autolock _l(mLock); 7300 ALOGV("setEnabled %p enabled %d", this, enabled); 7301 7302 if (enabled != isEnabled()) { 7303 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7304 if (enabled && status != NO_ERROR) { 7305 return status; 7306 } 7307 7308 switch (mState) { 7309 // going from disabled to enabled 7310 case IDLE: 7311 mState = STARTING; 7312 break; 7313 case STOPPED: 7314 mState = RESTART; 7315 break; 7316 case STOPPING: 7317 mState = ACTIVE; 7318 break; 7319 7320 // going from enabled to disabled 7321 case RESTART: 7322 mState = STOPPED; 7323 break; 7324 case STARTING: 7325 mState = IDLE; 7326 break; 7327 case ACTIVE: 7328 mState = STOPPING; 7329 break; 7330 case DESTROYED: 7331 return NO_ERROR; // simply ignore as we are being destroyed 7332 } 7333 for (size_t i = 1; i < mHandles.size(); i++) { 7334 sp<EffectHandle> h = mHandles[i].promote(); 7335 if (h != 0) { 7336 h->setEnabled(enabled); 7337 } 7338 } 7339 } 7340 return NO_ERROR; 7341} 7342 7343bool AudioFlinger::EffectModule::isEnabled() const 7344{ 7345 switch (mState) { 7346 case RESTART: 7347 case STARTING: 7348 case ACTIVE: 7349 return true; 7350 case IDLE: 7351 case STOPPING: 7352 case STOPPED: 7353 case DESTROYED: 7354 default: 7355 return false; 7356 } 7357} 7358 7359bool AudioFlinger::EffectModule::isProcessEnabled() const 7360{ 7361 switch (mState) { 7362 case RESTART: 7363 case ACTIVE: 7364 case STOPPING: 7365 case STOPPED: 7366 return true; 7367 case IDLE: 7368 case STARTING: 7369 case DESTROYED: 7370 default: 7371 return false; 7372 } 7373} 7374 7375status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7376{ 7377 Mutex::Autolock _l(mLock); 7378 status_t status = NO_ERROR; 7379 7380 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7381 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7382 if (isProcessEnabled() && 7383 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7384 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7385 status_t cmdStatus; 7386 uint32_t volume[2]; 7387 uint32_t *pVolume = NULL; 7388 uint32_t size = sizeof(volume); 7389 volume[0] = *left; 7390 volume[1] = *right; 7391 if (controller) { 7392 pVolume = volume; 7393 } 7394 status = (*mEffectInterface)->command(mEffectInterface, 7395 EFFECT_CMD_SET_VOLUME, 7396 size, 7397 volume, 7398 &size, 7399 pVolume); 7400 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7401 *left = volume[0]; 7402 *right = volume[1]; 7403 } 7404 } 7405 return status; 7406} 7407 7408status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7409{ 7410 Mutex::Autolock _l(mLock); 7411 status_t status = NO_ERROR; 7412 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7413 // audio pre processing modules on RecordThread can receive both output and 7414 // input device indication in the same call 7415 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7416 if (dev) { 7417 status_t cmdStatus; 7418 uint32_t size = sizeof(status_t); 7419 7420 status = (*mEffectInterface)->command(mEffectInterface, 7421 EFFECT_CMD_SET_DEVICE, 7422 sizeof(uint32_t), 7423 &dev, 7424 &size, 7425 &cmdStatus); 7426 if (status == NO_ERROR) { 7427 status = cmdStatus; 7428 } 7429 } 7430 dev = device & AUDIO_DEVICE_IN_ALL; 7431 if (dev) { 7432 status_t cmdStatus; 7433 uint32_t size = sizeof(status_t); 7434 7435 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7436 EFFECT_CMD_SET_INPUT_DEVICE, 7437 sizeof(uint32_t), 7438 &dev, 7439 &size, 7440 &cmdStatus); 7441 if (status2 == NO_ERROR) { 7442 status2 = cmdStatus; 7443 } 7444 if (status == NO_ERROR) { 7445 status = status2; 7446 } 7447 } 7448 } 7449 return status; 7450} 7451 7452status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7453{ 7454 Mutex::Autolock _l(mLock); 7455 status_t status = NO_ERROR; 7456 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7457 status_t cmdStatus; 7458 uint32_t size = sizeof(status_t); 7459 status = (*mEffectInterface)->command(mEffectInterface, 7460 EFFECT_CMD_SET_AUDIO_MODE, 7461 sizeof(audio_mode_t), 7462 &mode, 7463 &size, 7464 &cmdStatus); 7465 if (status == NO_ERROR) { 7466 status = cmdStatus; 7467 } 7468 } 7469 return status; 7470} 7471 7472void AudioFlinger::EffectModule::setSuspended(bool suspended) 7473{ 7474 Mutex::Autolock _l(mLock); 7475 mSuspended = suspended; 7476} 7477 7478bool AudioFlinger::EffectModule::suspended() const 7479{ 7480 Mutex::Autolock _l(mLock); 7481 return mSuspended; 7482} 7483 7484status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7485{ 7486 const size_t SIZE = 256; 7487 char buffer[SIZE]; 7488 String8 result; 7489 7490 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7491 result.append(buffer); 7492 7493 bool locked = tryLock(mLock); 7494 // failed to lock - AudioFlinger is probably deadlocked 7495 if (!locked) { 7496 result.append("\t\tCould not lock Fx mutex:\n"); 7497 } 7498 7499 result.append("\t\tSession Status State Engine:\n"); 7500 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7501 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7502 result.append(buffer); 7503 7504 result.append("\t\tDescriptor:\n"); 7505 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7506 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7507 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7508 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7509 result.append(buffer); 7510 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7511 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7512 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7513 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7514 result.append(buffer); 7515 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7516 mDescriptor.apiVersion, 7517 mDescriptor.flags); 7518 result.append(buffer); 7519 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7520 mDescriptor.name); 7521 result.append(buffer); 7522 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7523 mDescriptor.implementor); 7524 result.append(buffer); 7525 7526 result.append("\t\t- Input configuration:\n"); 7527 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7528 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7529 (uint32_t)mConfig.inputCfg.buffer.raw, 7530 mConfig.inputCfg.buffer.frameCount, 7531 mConfig.inputCfg.samplingRate, 7532 mConfig.inputCfg.channels, 7533 mConfig.inputCfg.format); 7534 result.append(buffer); 7535 7536 result.append("\t\t- Output configuration:\n"); 7537 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7538 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7539 (uint32_t)mConfig.outputCfg.buffer.raw, 7540 mConfig.outputCfg.buffer.frameCount, 7541 mConfig.outputCfg.samplingRate, 7542 mConfig.outputCfg.channels, 7543 mConfig.outputCfg.format); 7544 result.append(buffer); 7545 7546 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7547 result.append(buffer); 7548 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7549 for (size_t i = 0; i < mHandles.size(); ++i) { 7550 sp<EffectHandle> handle = mHandles[i].promote(); 7551 if (handle != 0) { 7552 handle->dump(buffer, SIZE); 7553 result.append(buffer); 7554 } 7555 } 7556 7557 result.append("\n"); 7558 7559 write(fd, result.string(), result.length()); 7560 7561 if (locked) { 7562 mLock.unlock(); 7563 } 7564 7565 return NO_ERROR; 7566} 7567 7568// ---------------------------------------------------------------------------- 7569// EffectHandle implementation 7570// ---------------------------------------------------------------------------- 7571 7572#undef LOG_TAG 7573#define LOG_TAG "AudioFlinger::EffectHandle" 7574 7575AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7576 const sp<AudioFlinger::Client>& client, 7577 const sp<IEffectClient>& effectClient, 7578 int32_t priority) 7579 : BnEffect(), 7580 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7581 mPriority(priority), mHasControl(false), mEnabled(false) 7582{ 7583 ALOGV("constructor %p", this); 7584 7585 if (client == 0) { 7586 return; 7587 } 7588 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7589 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7590 if (mCblkMemory != 0) { 7591 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7592 7593 if (mCblk != NULL) { 7594 new(mCblk) effect_param_cblk_t(); 7595 mBuffer = (uint8_t *)mCblk + bufOffset; 7596 } 7597 } else { 7598 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7599 return; 7600 } 7601} 7602 7603AudioFlinger::EffectHandle::~EffectHandle() 7604{ 7605 ALOGV("Destructor %p", this); 7606 disconnect(false); 7607 ALOGV("Destructor DONE %p", this); 7608} 7609 7610status_t AudioFlinger::EffectHandle::enable() 7611{ 7612 ALOGV("enable %p", this); 7613 if (!mHasControl) return INVALID_OPERATION; 7614 if (mEffect == 0) return DEAD_OBJECT; 7615 7616 if (mEnabled) { 7617 return NO_ERROR; 7618 } 7619 7620 mEnabled = true; 7621 7622 sp<ThreadBase> thread = mEffect->thread().promote(); 7623 if (thread != 0) { 7624 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7625 } 7626 7627 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7628 if (mEffect->suspended()) { 7629 return NO_ERROR; 7630 } 7631 7632 status_t status = mEffect->setEnabled(true); 7633 if (status != NO_ERROR) { 7634 if (thread != 0) { 7635 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7636 } 7637 mEnabled = false; 7638 } 7639 return status; 7640} 7641 7642status_t AudioFlinger::EffectHandle::disable() 7643{ 7644 ALOGV("disable %p", this); 7645 if (!mHasControl) return INVALID_OPERATION; 7646 if (mEffect == 0) return DEAD_OBJECT; 7647 7648 if (!mEnabled) { 7649 return NO_ERROR; 7650 } 7651 mEnabled = false; 7652 7653 if (mEffect->suspended()) { 7654 return NO_ERROR; 7655 } 7656 7657 status_t status = mEffect->setEnabled(false); 7658 7659 sp<ThreadBase> thread = mEffect->thread().promote(); 7660 if (thread != 0) { 7661 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7662 } 7663 7664 return status; 7665} 7666 7667void AudioFlinger::EffectHandle::disconnect() 7668{ 7669 disconnect(true); 7670} 7671 7672void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7673{ 7674 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7675 if (mEffect == 0) { 7676 return; 7677 } 7678 mEffect->disconnect(this, unpinIfLast); 7679 7680 if (mHasControl && mEnabled) { 7681 sp<ThreadBase> thread = mEffect->thread().promote(); 7682 if (thread != 0) { 7683 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7684 } 7685 } 7686 7687 // release sp on module => module destructor can be called now 7688 mEffect.clear(); 7689 if (mClient != 0) { 7690 if (mCblk != NULL) { 7691 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7692 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7693 } 7694 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7695 // Client destructor must run with AudioFlinger mutex locked 7696 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7697 mClient.clear(); 7698 } 7699} 7700 7701status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7702 uint32_t cmdSize, 7703 void *pCmdData, 7704 uint32_t *replySize, 7705 void *pReplyData) 7706{ 7707// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7708// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7709 7710 // only get parameter command is permitted for applications not controlling the effect 7711 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7712 return INVALID_OPERATION; 7713 } 7714 if (mEffect == 0) return DEAD_OBJECT; 7715 if (mClient == 0) return INVALID_OPERATION; 7716 7717 // handle commands that are not forwarded transparently to effect engine 7718 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7719 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7720 // no risk to block the whole media server process or mixer threads is we are stuck here 7721 Mutex::Autolock _l(mCblk->lock); 7722 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7723 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7724 mCblk->serverIndex = 0; 7725 mCblk->clientIndex = 0; 7726 return BAD_VALUE; 7727 } 7728 status_t status = NO_ERROR; 7729 while (mCblk->serverIndex < mCblk->clientIndex) { 7730 int reply; 7731 uint32_t rsize = sizeof(int); 7732 int *p = (int *)(mBuffer + mCblk->serverIndex); 7733 int size = *p++; 7734 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7735 ALOGW("command(): invalid parameter block size"); 7736 break; 7737 } 7738 effect_param_t *param = (effect_param_t *)p; 7739 if (param->psize == 0 || param->vsize == 0) { 7740 ALOGW("command(): null parameter or value size"); 7741 mCblk->serverIndex += size; 7742 continue; 7743 } 7744 uint32_t psize = sizeof(effect_param_t) + 7745 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7746 param->vsize; 7747 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7748 psize, 7749 p, 7750 &rsize, 7751 &reply); 7752 // stop at first error encountered 7753 if (ret != NO_ERROR) { 7754 status = ret; 7755 *(int *)pReplyData = reply; 7756 break; 7757 } else if (reply != NO_ERROR) { 7758 *(int *)pReplyData = reply; 7759 break; 7760 } 7761 mCblk->serverIndex += size; 7762 } 7763 mCblk->serverIndex = 0; 7764 mCblk->clientIndex = 0; 7765 return status; 7766 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7767 *(int *)pReplyData = NO_ERROR; 7768 return enable(); 7769 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7770 *(int *)pReplyData = NO_ERROR; 7771 return disable(); 7772 } 7773 7774 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7775} 7776 7777void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7778{ 7779 ALOGV("setControl %p control %d", this, hasControl); 7780 7781 mHasControl = hasControl; 7782 mEnabled = enabled; 7783 7784 if (signal && mEffectClient != 0) { 7785 mEffectClient->controlStatusChanged(hasControl); 7786 } 7787} 7788 7789void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7790 uint32_t cmdSize, 7791 void *pCmdData, 7792 uint32_t replySize, 7793 void *pReplyData) 7794{ 7795 if (mEffectClient != 0) { 7796 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7797 } 7798} 7799 7800 7801 7802void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7803{ 7804 if (mEffectClient != 0) { 7805 mEffectClient->enableStatusChanged(enabled); 7806 } 7807} 7808 7809status_t AudioFlinger::EffectHandle::onTransact( 7810 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7811{ 7812 return BnEffect::onTransact(code, data, reply, flags); 7813} 7814 7815 7816void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7817{ 7818 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7819 7820 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7821 (mClient == 0) ? getpid_cached : mClient->pid(), 7822 mPriority, 7823 mHasControl, 7824 !locked, 7825 mCblk ? mCblk->clientIndex : 0, 7826 mCblk ? mCblk->serverIndex : 0 7827 ); 7828 7829 if (locked) { 7830 mCblk->lock.unlock(); 7831 } 7832} 7833 7834#undef LOG_TAG 7835#define LOG_TAG "AudioFlinger::EffectChain" 7836 7837AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7838 int sessionId) 7839 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7840 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7841 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7842{ 7843 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7844 if (thread == NULL) { 7845 return; 7846 } 7847 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7848 thread->frameCount(); 7849} 7850 7851AudioFlinger::EffectChain::~EffectChain() 7852{ 7853 if (mOwnInBuffer) { 7854 delete mInBuffer; 7855 } 7856 7857} 7858 7859// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7860sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7861{ 7862 size_t size = mEffects.size(); 7863 7864 for (size_t i = 0; i < size; i++) { 7865 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7866 return mEffects[i]; 7867 } 7868 } 7869 return 0; 7870} 7871 7872// getEffectFromId_l() must be called with ThreadBase::mLock held 7873sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7874{ 7875 size_t size = mEffects.size(); 7876 7877 for (size_t i = 0; i < size; i++) { 7878 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7879 if (id == 0 || mEffects[i]->id() == id) { 7880 return mEffects[i]; 7881 } 7882 } 7883 return 0; 7884} 7885 7886// getEffectFromType_l() must be called with ThreadBase::mLock held 7887sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7888 const effect_uuid_t *type) 7889{ 7890 size_t size = mEffects.size(); 7891 7892 for (size_t i = 0; i < size; i++) { 7893 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7894 return mEffects[i]; 7895 } 7896 } 7897 return 0; 7898} 7899 7900// Must be called with EffectChain::mLock locked 7901void AudioFlinger::EffectChain::process_l() 7902{ 7903 sp<ThreadBase> thread = mThread.promote(); 7904 if (thread == 0) { 7905 ALOGW("process_l(): cannot promote mixer thread"); 7906 return; 7907 } 7908 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7909 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7910 // always process effects unless no more tracks are on the session and the effect tail 7911 // has been rendered 7912 bool doProcess = true; 7913 if (!isGlobalSession) { 7914 bool tracksOnSession = (trackCnt() != 0); 7915 7916 if (!tracksOnSession && mTailBufferCount == 0) { 7917 doProcess = false; 7918 } 7919 7920 if (activeTrackCnt() == 0) { 7921 // if no track is active and the effect tail has not been rendered, 7922 // the input buffer must be cleared here as the mixer process will not do it 7923 if (tracksOnSession || mTailBufferCount > 0) { 7924 size_t numSamples = thread->frameCount() * thread->channelCount(); 7925 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7926 if (mTailBufferCount > 0) { 7927 mTailBufferCount--; 7928 } 7929 } 7930 } 7931 } 7932 7933 size_t size = mEffects.size(); 7934 if (doProcess) { 7935 for (size_t i = 0; i < size; i++) { 7936 mEffects[i]->process(); 7937 } 7938 } 7939 for (size_t i = 0; i < size; i++) { 7940 mEffects[i]->updateState(); 7941 } 7942} 7943 7944// addEffect_l() must be called with PlaybackThread::mLock held 7945status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7946{ 7947 effect_descriptor_t desc = effect->desc(); 7948 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7949 7950 Mutex::Autolock _l(mLock); 7951 effect->setChain(this); 7952 sp<ThreadBase> thread = mThread.promote(); 7953 if (thread == 0) { 7954 return NO_INIT; 7955 } 7956 effect->setThread(thread); 7957 7958 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7959 // Auxiliary effects are inserted at the beginning of mEffects vector as 7960 // they are processed first and accumulated in chain input buffer 7961 mEffects.insertAt(effect, 0); 7962 7963 // the input buffer for auxiliary effect contains mono samples in 7964 // 32 bit format. This is to avoid saturation in AudoMixer 7965 // accumulation stage. Saturation is done in EffectModule::process() before 7966 // calling the process in effect engine 7967 size_t numSamples = thread->frameCount(); 7968 int32_t *buffer = new int32_t[numSamples]; 7969 memset(buffer, 0, numSamples * sizeof(int32_t)); 7970 effect->setInBuffer((int16_t *)buffer); 7971 // auxiliary effects output samples to chain input buffer for further processing 7972 // by insert effects 7973 effect->setOutBuffer(mInBuffer); 7974 } else { 7975 // Insert effects are inserted at the end of mEffects vector as they are processed 7976 // after track and auxiliary effects. 7977 // Insert effect order as a function of indicated preference: 7978 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7979 // another effect is present 7980 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7981 // last effect claiming first position 7982 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7983 // first effect claiming last position 7984 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7985 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7986 // already present 7987 7988 size_t size = mEffects.size(); 7989 size_t idx_insert = size; 7990 ssize_t idx_insert_first = -1; 7991 ssize_t idx_insert_last = -1; 7992 7993 for (size_t i = 0; i < size; i++) { 7994 effect_descriptor_t d = mEffects[i]->desc(); 7995 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7996 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7997 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7998 // check invalid effect chaining combinations 7999 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8000 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8001 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8002 return INVALID_OPERATION; 8003 } 8004 // remember position of first insert effect and by default 8005 // select this as insert position for new effect 8006 if (idx_insert == size) { 8007 idx_insert = i; 8008 } 8009 // remember position of last insert effect claiming 8010 // first position 8011 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8012 idx_insert_first = i; 8013 } 8014 // remember position of first insert effect claiming 8015 // last position 8016 if (iPref == EFFECT_FLAG_INSERT_LAST && 8017 idx_insert_last == -1) { 8018 idx_insert_last = i; 8019 } 8020 } 8021 } 8022 8023 // modify idx_insert from first position if needed 8024 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8025 if (idx_insert_last != -1) { 8026 idx_insert = idx_insert_last; 8027 } else { 8028 idx_insert = size; 8029 } 8030 } else { 8031 if (idx_insert_first != -1) { 8032 idx_insert = idx_insert_first + 1; 8033 } 8034 } 8035 8036 // always read samples from chain input buffer 8037 effect->setInBuffer(mInBuffer); 8038 8039 // if last effect in the chain, output samples to chain 8040 // output buffer, otherwise to chain input buffer 8041 if (idx_insert == size) { 8042 if (idx_insert != 0) { 8043 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8044 mEffects[idx_insert-1]->configure(); 8045 } 8046 effect->setOutBuffer(mOutBuffer); 8047 } else { 8048 effect->setOutBuffer(mInBuffer); 8049 } 8050 mEffects.insertAt(effect, idx_insert); 8051 8052 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8053 } 8054 effect->configure(); 8055 return NO_ERROR; 8056} 8057 8058// removeEffect_l() must be called with PlaybackThread::mLock held 8059size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8060{ 8061 Mutex::Autolock _l(mLock); 8062 size_t size = mEffects.size(); 8063 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8064 8065 for (size_t i = 0; i < size; i++) { 8066 if (effect == mEffects[i]) { 8067 // calling stop here will remove pre-processing effect from the audio HAL. 8068 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8069 // the middle of a read from audio HAL 8070 if (mEffects[i]->state() == EffectModule::ACTIVE || 8071 mEffects[i]->state() == EffectModule::STOPPING) { 8072 mEffects[i]->stop(); 8073 } 8074 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8075 delete[] effect->inBuffer(); 8076 } else { 8077 if (i == size - 1 && i != 0) { 8078 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8079 mEffects[i - 1]->configure(); 8080 } 8081 } 8082 mEffects.removeAt(i); 8083 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8084 break; 8085 } 8086 } 8087 8088 return mEffects.size(); 8089} 8090 8091// setDevice_l() must be called with PlaybackThread::mLock held 8092void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8093{ 8094 size_t size = mEffects.size(); 8095 for (size_t i = 0; i < size; i++) { 8096 mEffects[i]->setDevice(device); 8097 } 8098} 8099 8100// setMode_l() must be called with PlaybackThread::mLock held 8101void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8102{ 8103 size_t size = mEffects.size(); 8104 for (size_t i = 0; i < size; i++) { 8105 mEffects[i]->setMode(mode); 8106 } 8107} 8108 8109// setVolume_l() must be called with PlaybackThread::mLock held 8110bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8111{ 8112 uint32_t newLeft = *left; 8113 uint32_t newRight = *right; 8114 bool hasControl = false; 8115 int ctrlIdx = -1; 8116 size_t size = mEffects.size(); 8117 8118 // first update volume controller 8119 for (size_t i = size; i > 0; i--) { 8120 if (mEffects[i - 1]->isProcessEnabled() && 8121 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8122 ctrlIdx = i - 1; 8123 hasControl = true; 8124 break; 8125 } 8126 } 8127 8128 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8129 if (hasControl) { 8130 *left = mNewLeftVolume; 8131 *right = mNewRightVolume; 8132 } 8133 return hasControl; 8134 } 8135 8136 mVolumeCtrlIdx = ctrlIdx; 8137 mLeftVolume = newLeft; 8138 mRightVolume = newRight; 8139 8140 // second get volume update from volume controller 8141 if (ctrlIdx >= 0) { 8142 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8143 mNewLeftVolume = newLeft; 8144 mNewRightVolume = newRight; 8145 } 8146 // then indicate volume to all other effects in chain. 8147 // Pass altered volume to effects before volume controller 8148 // and requested volume to effects after controller 8149 uint32_t lVol = newLeft; 8150 uint32_t rVol = newRight; 8151 8152 for (size_t i = 0; i < size; i++) { 8153 if ((int)i == ctrlIdx) continue; 8154 // this also works for ctrlIdx == -1 when there is no volume controller 8155 if ((int)i > ctrlIdx) { 8156 lVol = *left; 8157 rVol = *right; 8158 } 8159 mEffects[i]->setVolume(&lVol, &rVol, false); 8160 } 8161 *left = newLeft; 8162 *right = newRight; 8163 8164 return hasControl; 8165} 8166 8167status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8168{ 8169 const size_t SIZE = 256; 8170 char buffer[SIZE]; 8171 String8 result; 8172 8173 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8174 result.append(buffer); 8175 8176 bool locked = tryLock(mLock); 8177 // failed to lock - AudioFlinger is probably deadlocked 8178 if (!locked) { 8179 result.append("\tCould not lock mutex:\n"); 8180 } 8181 8182 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8183 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8184 mEffects.size(), 8185 (uint32_t)mInBuffer, 8186 (uint32_t)mOutBuffer, 8187 mActiveTrackCnt); 8188 result.append(buffer); 8189 write(fd, result.string(), result.size()); 8190 8191 for (size_t i = 0; i < mEffects.size(); ++i) { 8192 sp<EffectModule> effect = mEffects[i]; 8193 if (effect != 0) { 8194 effect->dump(fd, args); 8195 } 8196 } 8197 8198 if (locked) { 8199 mLock.unlock(); 8200 } 8201 8202 return NO_ERROR; 8203} 8204 8205// must be called with ThreadBase::mLock held 8206void AudioFlinger::EffectChain::setEffectSuspended_l( 8207 const effect_uuid_t *type, bool suspend) 8208{ 8209 sp<SuspendedEffectDesc> desc; 8210 // use effect type UUID timelow as key as there is no real risk of identical 8211 // timeLow fields among effect type UUIDs. 8212 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8213 if (suspend) { 8214 if (index >= 0) { 8215 desc = mSuspendedEffects.valueAt(index); 8216 } else { 8217 desc = new SuspendedEffectDesc(); 8218 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8219 mSuspendedEffects.add(type->timeLow, desc); 8220 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8221 } 8222 if (desc->mRefCount++ == 0) { 8223 sp<EffectModule> effect = getEffectIfEnabled(type); 8224 if (effect != 0) { 8225 desc->mEffect = effect; 8226 effect->setSuspended(true); 8227 effect->setEnabled(false); 8228 } 8229 } 8230 } else { 8231 if (index < 0) { 8232 return; 8233 } 8234 desc = mSuspendedEffects.valueAt(index); 8235 if (desc->mRefCount <= 0) { 8236 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8237 desc->mRefCount = 1; 8238 } 8239 if (--desc->mRefCount == 0) { 8240 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8241 if (desc->mEffect != 0) { 8242 sp<EffectModule> effect = desc->mEffect.promote(); 8243 if (effect != 0) { 8244 effect->setSuspended(false); 8245 sp<EffectHandle> handle = effect->controlHandle(); 8246 if (handle != 0) { 8247 effect->setEnabled(handle->enabled()); 8248 } 8249 } 8250 desc->mEffect.clear(); 8251 } 8252 mSuspendedEffects.removeItemsAt(index); 8253 } 8254 } 8255} 8256 8257// must be called with ThreadBase::mLock held 8258void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8259{ 8260 sp<SuspendedEffectDesc> desc; 8261 8262 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8263 if (suspend) { 8264 if (index >= 0) { 8265 desc = mSuspendedEffects.valueAt(index); 8266 } else { 8267 desc = new SuspendedEffectDesc(); 8268 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8269 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8270 } 8271 if (desc->mRefCount++ == 0) { 8272 Vector< sp<EffectModule> > effects; 8273 getSuspendEligibleEffects(effects); 8274 for (size_t i = 0; i < effects.size(); i++) { 8275 setEffectSuspended_l(&effects[i]->desc().type, true); 8276 } 8277 } 8278 } else { 8279 if (index < 0) { 8280 return; 8281 } 8282 desc = mSuspendedEffects.valueAt(index); 8283 if (desc->mRefCount <= 0) { 8284 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8285 desc->mRefCount = 1; 8286 } 8287 if (--desc->mRefCount == 0) { 8288 Vector<const effect_uuid_t *> types; 8289 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8290 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8291 continue; 8292 } 8293 types.add(&mSuspendedEffects.valueAt(i)->mType); 8294 } 8295 for (size_t i = 0; i < types.size(); i++) { 8296 setEffectSuspended_l(types[i], false); 8297 } 8298 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8299 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8300 } 8301 } 8302} 8303 8304 8305// The volume effect is used for automated tests only 8306#ifndef OPENSL_ES_H_ 8307static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8308 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8309const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8310#endif //OPENSL_ES_H_ 8311 8312bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8313{ 8314 // auxiliary effects and visualizer are never suspended on output mix 8315 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8316 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8317 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8318 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8319 return false; 8320 } 8321 return true; 8322} 8323 8324void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8325{ 8326 effects.clear(); 8327 for (size_t i = 0; i < mEffects.size(); i++) { 8328 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8329 effects.add(mEffects[i]); 8330 } 8331 } 8332} 8333 8334sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8335 const effect_uuid_t *type) 8336{ 8337 sp<EffectModule> effect = getEffectFromType_l(type); 8338 return effect != 0 && effect->isEnabled() ? effect : 0; 8339} 8340 8341void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8342 bool enabled) 8343{ 8344 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8345 if (enabled) { 8346 if (index < 0) { 8347 // if the effect is not suspend check if all effects are suspended 8348 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8349 if (index < 0) { 8350 return; 8351 } 8352 if (!isEffectEligibleForSuspend(effect->desc())) { 8353 return; 8354 } 8355 setEffectSuspended_l(&effect->desc().type, enabled); 8356 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8357 if (index < 0) { 8358 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8359 return; 8360 } 8361 } 8362 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8363 effect->desc().type.timeLow); 8364 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8365 // if effect is requested to suspended but was not yet enabled, supend it now. 8366 if (desc->mEffect == 0) { 8367 desc->mEffect = effect; 8368 effect->setEnabled(false); 8369 effect->setSuspended(true); 8370 } 8371 } else { 8372 if (index < 0) { 8373 return; 8374 } 8375 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8376 effect->desc().type.timeLow); 8377 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8378 desc->mEffect.clear(); 8379 effect->setSuspended(false); 8380 } 8381} 8382 8383#undef LOG_TAG 8384#define LOG_TAG "AudioFlinger" 8385 8386// ---------------------------------------------------------------------------- 8387 8388status_t AudioFlinger::onTransact( 8389 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8390{ 8391 return BnAudioFlinger::onTransact(code, data, reply, flags); 8392} 8393 8394}; // namespace android 8395