AudioFlinger.cpp revision 73d227557ba5192735356bacab9f77b44980793b
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 "primary", 154 "a2dp", 155 "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 // initialized by stream_type_t default constructor 1507 // mStreamTypes[stream].valid = true; 1508 } 1509 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1510 // because mAudioFlinger doesn't have one to copy from 1511} 1512 1513AudioFlinger::PlaybackThread::~PlaybackThread() 1514{ 1515 delete [] mMixBuffer; 1516} 1517 1518status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1519{ 1520 dumpInternals(fd, args); 1521 dumpTracks(fd, args); 1522 dumpEffectChains(fd, args); 1523 return NO_ERROR; 1524} 1525 1526status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1527{ 1528 const size_t SIZE = 256; 1529 char buffer[SIZE]; 1530 String8 result; 1531 1532 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1533 result.append(buffer); 1534 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1535 for (size_t i = 0; i < mTracks.size(); ++i) { 1536 sp<Track> track = mTracks[i]; 1537 if (track != 0) { 1538 track->dump(buffer, SIZE); 1539 result.append(buffer); 1540 } 1541 } 1542 1543 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1544 result.append(buffer); 1545 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1546 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1547 sp<Track> track = mActiveTracks[i].promote(); 1548 if (track != 0) { 1549 track->dump(buffer, SIZE); 1550 result.append(buffer); 1551 } 1552 } 1553 write(fd, result.string(), result.size()); 1554 return NO_ERROR; 1555} 1556 1557status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1558{ 1559 const size_t SIZE = 256; 1560 char buffer[SIZE]; 1561 String8 result; 1562 1563 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1564 result.append(buffer); 1565 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1566 result.append(buffer); 1567 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1568 result.append(buffer); 1569 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1570 result.append(buffer); 1571 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1572 result.append(buffer); 1573 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1574 result.append(buffer); 1575 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1576 result.append(buffer); 1577 write(fd, result.string(), result.size()); 1578 1579 dumpBase(fd, args); 1580 1581 return NO_ERROR; 1582} 1583 1584// Thread virtuals 1585status_t AudioFlinger::PlaybackThread::readyToRun() 1586{ 1587 status_t status = initCheck(); 1588 if (status == NO_ERROR) { 1589 ALOGI("AudioFlinger's thread %p ready to run", this); 1590 } else { 1591 ALOGE("No working audio driver found."); 1592 } 1593 return status; 1594} 1595 1596void AudioFlinger::PlaybackThread::onFirstRef() 1597{ 1598 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1599} 1600 1601// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1602sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1603 const sp<AudioFlinger::Client>& client, 1604 audio_stream_type_t streamType, 1605 uint32_t sampleRate, 1606 audio_format_t format, 1607 uint32_t channelMask, 1608 int frameCount, 1609 const sp<IMemory>& sharedBuffer, 1610 int sessionId, 1611 IAudioFlinger::track_flags_t flags, 1612 status_t *status) 1613{ 1614 sp<Track> track; 1615 status_t lStatus; 1616 1617 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1618 1619 // client expresses a preference for FAST, but we get the final say 1620 if ((flags & IAudioFlinger::TRACK_FAST) && 1621 !( 1622 // not timed 1623 (!isTimed) && 1624 // either of these use cases: 1625 ( 1626 // use case 1: shared buffer with any frame count 1627 ( 1628 (sharedBuffer != 0) 1629 ) || 1630 // use case 2: callback handler and small power-of-2 frame count 1631 ( 1632 // unfortunately we can't verify that there's a callback until start() 1633 // FIXME supported frame counts should not be hard-coded 1634 ( 1635 (frameCount == 128) || 1636 (frameCount == 256) || 1637 (frameCount == 512) 1638 ) 1639 ) 1640 ) && 1641 // PCM data 1642 audio_is_linear_pcm(format) && 1643 // mono or stereo 1644 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1645 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1646 // hardware sample rate 1647 (sampleRate == mSampleRate) 1648 // FIXME test that MixerThread for this fast track has a capable output HAL 1649 // FIXME add a permission test also? 1650 ) ) { 1651 ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied"); 1652 flags &= ~IAudioFlinger::TRACK_FAST; 1653 } 1654 1655 if (mType == DIRECT) { 1656 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1657 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1658 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1659 "for output %p with format %d", 1660 sampleRate, format, channelMask, mOutput, mFormat); 1661 lStatus = BAD_VALUE; 1662 goto Exit; 1663 } 1664 } 1665 } else { 1666 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1667 if (sampleRate > mSampleRate*2) { 1668 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1669 lStatus = BAD_VALUE; 1670 goto Exit; 1671 } 1672 } 1673 1674 lStatus = initCheck(); 1675 if (lStatus != NO_ERROR) { 1676 ALOGE("Audio driver not initialized."); 1677 goto Exit; 1678 } 1679 1680 { // scope for mLock 1681 Mutex::Autolock _l(mLock); 1682 1683 // all tracks in same audio session must share the same routing strategy otherwise 1684 // conflicts will happen when tracks are moved from one output to another by audio policy 1685 // manager 1686 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1687 for (size_t i = 0; i < mTracks.size(); ++i) { 1688 sp<Track> t = mTracks[i]; 1689 if (t != 0 && !t->isOutputTrack()) { 1690 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1691 if (sessionId == t->sessionId() && strategy != actual) { 1692 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1693 strategy, actual); 1694 lStatus = BAD_VALUE; 1695 goto Exit; 1696 } 1697 } 1698 } 1699 1700 if (!isTimed) { 1701 track = new Track(this, client, streamType, sampleRate, format, 1702 channelMask, frameCount, sharedBuffer, sessionId, flags); 1703 } else { 1704 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1705 channelMask, frameCount, sharedBuffer, sessionId); 1706 } 1707 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1708 lStatus = NO_MEMORY; 1709 goto Exit; 1710 } 1711 mTracks.add(track); 1712 1713 sp<EffectChain> chain = getEffectChain_l(sessionId); 1714 if (chain != 0) { 1715 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1716 track->setMainBuffer(chain->inBuffer()); 1717 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1718 chain->incTrackCnt(); 1719 } 1720 1721 // invalidate track immediately if the stream type was moved to another thread since 1722 // createTrack() was called by the client process. 1723 if (!mStreamTypes[streamType].valid) { 1724 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1725 this, streamType); 1726 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1727 } 1728 } 1729 lStatus = NO_ERROR; 1730 1731Exit: 1732 if (status) { 1733 *status = lStatus; 1734 } 1735 return track; 1736} 1737 1738uint32_t AudioFlinger::PlaybackThread::latency() const 1739{ 1740 Mutex::Autolock _l(mLock); 1741 if (initCheck() == NO_ERROR) { 1742 return mOutput->stream->get_latency(mOutput->stream); 1743 } else { 1744 return 0; 1745 } 1746} 1747 1748void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1749{ 1750 Mutex::Autolock _l(mLock); 1751 mMasterVolume = value; 1752} 1753 1754void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1755{ 1756 Mutex::Autolock _l(mLock); 1757 setMasterMute_l(muted); 1758} 1759 1760void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1761{ 1762 Mutex::Autolock _l(mLock); 1763 mStreamTypes[stream].volume = value; 1764} 1765 1766void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1767{ 1768 Mutex::Autolock _l(mLock); 1769 mStreamTypes[stream].mute = muted; 1770} 1771 1772float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1773{ 1774 Mutex::Autolock _l(mLock); 1775 return mStreamTypes[stream].volume; 1776} 1777 1778// addTrack_l() must be called with ThreadBase::mLock held 1779status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1780{ 1781 status_t status = ALREADY_EXISTS; 1782 1783 // set retry count for buffer fill 1784 track->mRetryCount = kMaxTrackStartupRetries; 1785 if (mActiveTracks.indexOf(track) < 0) { 1786 // the track is newly added, make sure it fills up all its 1787 // buffers before playing. This is to ensure the client will 1788 // effectively get the latency it requested. 1789 track->mFillingUpStatus = Track::FS_FILLING; 1790 track->mResetDone = false; 1791 mActiveTracks.add(track); 1792 if (track->mainBuffer() != mMixBuffer) { 1793 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1794 if (chain != 0) { 1795 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1796 chain->incActiveTrackCnt(); 1797 } 1798 } 1799 1800 status = NO_ERROR; 1801 } 1802 1803 ALOGV("mWaitWorkCV.broadcast"); 1804 mWaitWorkCV.broadcast(); 1805 1806 return status; 1807} 1808 1809// destroyTrack_l() must be called with ThreadBase::mLock held 1810void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1811{ 1812 track->mState = TrackBase::TERMINATED; 1813 if (mActiveTracks.indexOf(track) < 0) { 1814 removeTrack_l(track); 1815 } 1816} 1817 1818void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1819{ 1820 mTracks.remove(track); 1821 deleteTrackName_l(track->name()); 1822 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1823 if (chain != 0) { 1824 chain->decTrackCnt(); 1825 } 1826} 1827 1828String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1829{ 1830 String8 out_s8 = String8(""); 1831 char *s; 1832 1833 Mutex::Autolock _l(mLock); 1834 if (initCheck() != NO_ERROR) { 1835 return out_s8; 1836 } 1837 1838 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1839 out_s8 = String8(s); 1840 free(s); 1841 return out_s8; 1842} 1843 1844// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1845void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1846 AudioSystem::OutputDescriptor desc; 1847 void *param2 = NULL; 1848 1849 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1850 1851 switch (event) { 1852 case AudioSystem::OUTPUT_OPENED: 1853 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1854 desc.channels = mChannelMask; 1855 desc.samplingRate = mSampleRate; 1856 desc.format = mFormat; 1857 desc.frameCount = mFrameCount; 1858 desc.latency = latency(); 1859 param2 = &desc; 1860 break; 1861 1862 case AudioSystem::STREAM_CONFIG_CHANGED: 1863 param2 = ¶m; 1864 case AudioSystem::OUTPUT_CLOSED: 1865 default: 1866 break; 1867 } 1868 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1869} 1870 1871void AudioFlinger::PlaybackThread::readOutputParameters() 1872{ 1873 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1874 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1875 mChannelCount = (uint16_t)popcount(mChannelMask); 1876 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1877 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1878 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1879 1880 // FIXME - Current mixer implementation only supports stereo output: Always 1881 // Allocate a stereo buffer even if HW output is mono. 1882 delete[] mMixBuffer; 1883 mMixBuffer = new int16_t[mFrameCount * 2]; 1884 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1885 1886 // force reconfiguration of effect chains and engines to take new buffer size and audio 1887 // parameters into account 1888 // Note that mLock is not held when readOutputParameters() is called from the constructor 1889 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1890 // matter. 1891 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1892 Vector< sp<EffectChain> > effectChains = mEffectChains; 1893 for (size_t i = 0; i < effectChains.size(); i ++) { 1894 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1895 } 1896} 1897 1898status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1899{ 1900 if (halFrames == NULL || dspFrames == NULL) { 1901 return BAD_VALUE; 1902 } 1903 Mutex::Autolock _l(mLock); 1904 if (initCheck() != NO_ERROR) { 1905 return INVALID_OPERATION; 1906 } 1907 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1908 1909 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1910} 1911 1912uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1913{ 1914 Mutex::Autolock _l(mLock); 1915 uint32_t result = 0; 1916 if (getEffectChain_l(sessionId) != 0) { 1917 result = EFFECT_SESSION; 1918 } 1919 1920 for (size_t i = 0; i < mTracks.size(); ++i) { 1921 sp<Track> track = mTracks[i]; 1922 if (sessionId == track->sessionId() && 1923 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1924 result |= TRACK_SESSION; 1925 break; 1926 } 1927 } 1928 1929 return result; 1930} 1931 1932uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1933{ 1934 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1935 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1936 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1937 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1938 } 1939 for (size_t i = 0; i < mTracks.size(); i++) { 1940 sp<Track> track = mTracks[i]; 1941 if (sessionId == track->sessionId() && 1942 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1943 return AudioSystem::getStrategyForStream(track->streamType()); 1944 } 1945 } 1946 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1947} 1948 1949 1950AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1951{ 1952 Mutex::Autolock _l(mLock); 1953 return mOutput; 1954} 1955 1956AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1957{ 1958 Mutex::Autolock _l(mLock); 1959 AudioStreamOut *output = mOutput; 1960 mOutput = NULL; 1961 return output; 1962} 1963 1964// this method must always be called either with ThreadBase mLock held or inside the thread loop 1965audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1966{ 1967 if (mOutput == NULL) { 1968 return NULL; 1969 } 1970 return &mOutput->stream->common; 1971} 1972 1973uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1974{ 1975 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1976 // decoding and transfer time. So sleeping for half of the latency would likely cause 1977 // underruns 1978 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1979 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1980 } else { 1981 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1982 } 1983} 1984 1985status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1986{ 1987 if (!isValidSyncEvent(event)) { 1988 return BAD_VALUE; 1989 } 1990 1991 Mutex::Autolock _l(mLock); 1992 1993 for (size_t i = 0; i < mTracks.size(); ++i) { 1994 sp<Track> track = mTracks[i]; 1995 if (event->triggerSession() == track->sessionId()) { 1996 track->setSyncEvent(event); 1997 return NO_ERROR; 1998 } 1999 } 2000 2001 return NAME_NOT_FOUND; 2002} 2003 2004bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2005{ 2006 switch (event->type()) { 2007 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2008 return true; 2009 default: 2010 break; 2011 } 2012 return false; 2013} 2014 2015// ---------------------------------------------------------------------------- 2016 2017AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2018 audio_io_handle_t id, uint32_t device, type_t type) 2019 : PlaybackThread(audioFlinger, output, id, device, type) 2020{ 2021 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2022 // FIXME - Current mixer implementation only supports stereo output 2023 if (mChannelCount == 1) { 2024 ALOGE("Invalid audio hardware channel count"); 2025 } 2026} 2027 2028AudioFlinger::MixerThread::~MixerThread() 2029{ 2030 delete mAudioMixer; 2031} 2032 2033class CpuStats { 2034public: 2035 CpuStats(); 2036 void sample(const String8 &title); 2037#ifdef DEBUG_CPU_USAGE 2038private: 2039 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2040 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2041 2042 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2043 2044 int mCpuNum; // thread's current CPU number 2045 int mCpukHz; // frequency of thread's current CPU in kHz 2046#endif 2047}; 2048 2049CpuStats::CpuStats() 2050#ifdef DEBUG_CPU_USAGE 2051 : mCpuNum(-1), mCpukHz(-1) 2052#endif 2053{ 2054} 2055 2056void CpuStats::sample(const String8 &title) { 2057#ifdef DEBUG_CPU_USAGE 2058 // get current thread's delta CPU time in wall clock ns 2059 double wcNs; 2060 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2061 2062 // record sample for wall clock statistics 2063 if (valid) { 2064 mWcStats.sample(wcNs); 2065 } 2066 2067 // get the current CPU number 2068 int cpuNum = sched_getcpu(); 2069 2070 // get the current CPU frequency in kHz 2071 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2072 2073 // check if either CPU number or frequency changed 2074 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2075 mCpuNum = cpuNum; 2076 mCpukHz = cpukHz; 2077 // ignore sample for purposes of cycles 2078 valid = false; 2079 } 2080 2081 // if no change in CPU number or frequency, then record sample for cycle statistics 2082 if (valid && mCpukHz > 0) { 2083 double cycles = wcNs * cpukHz * 0.000001; 2084 mHzStats.sample(cycles); 2085 } 2086 2087 unsigned n = mWcStats.n(); 2088 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2089 if ((n & 127) == 1) { 2090 long long elapsed = mCpuUsage.elapsed(); 2091 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2092 double perLoop = elapsed / (double) n; 2093 double perLoop100 = perLoop * 0.01; 2094 double perLoop1k = perLoop * 0.001; 2095 double mean = mWcStats.mean(); 2096 double stddev = mWcStats.stddev(); 2097 double minimum = mWcStats.minimum(); 2098 double maximum = mWcStats.maximum(); 2099 double meanCycles = mHzStats.mean(); 2100 double stddevCycles = mHzStats.stddev(); 2101 double minCycles = mHzStats.minimum(); 2102 double maxCycles = mHzStats.maximum(); 2103 mCpuUsage.resetElapsed(); 2104 mWcStats.reset(); 2105 mHzStats.reset(); 2106 ALOGD("CPU usage for %s over past %.1f secs\n" 2107 " (%u mixer loops at %.1f mean ms per loop):\n" 2108 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2109 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2110 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2111 title.string(), 2112 elapsed * .000000001, n, perLoop * .000001, 2113 mean * .001, 2114 stddev * .001, 2115 minimum * .001, 2116 maximum * .001, 2117 mean / perLoop100, 2118 stddev / perLoop100, 2119 minimum / perLoop100, 2120 maximum / perLoop100, 2121 meanCycles / perLoop1k, 2122 stddevCycles / perLoop1k, 2123 minCycles / perLoop1k, 2124 maxCycles / perLoop1k); 2125 2126 } 2127 } 2128#endif 2129}; 2130 2131void AudioFlinger::PlaybackThread::checkSilentMode_l() 2132{ 2133 if (!mMasterMute) { 2134 char value[PROPERTY_VALUE_MAX]; 2135 if (property_get("ro.audio.silent", value, "0") > 0) { 2136 char *endptr; 2137 unsigned long ul = strtoul(value, &endptr, 0); 2138 if (*endptr == '\0' && ul != 0) { 2139 ALOGD("Silence is golden"); 2140 // The setprop command will not allow a property to be changed after 2141 // the first time it is set, so we don't have to worry about un-muting. 2142 setMasterMute_l(true); 2143 } 2144 } 2145 } 2146} 2147 2148bool AudioFlinger::PlaybackThread::threadLoop() 2149{ 2150 Vector< sp<Track> > tracksToRemove; 2151 2152 standbyTime = systemTime(); 2153 2154 // MIXER 2155 nsecs_t lastWarning = 0; 2156if (mType == MIXER) { 2157 longStandbyExit = false; 2158} 2159 2160 // DUPLICATING 2161 // FIXME could this be made local to while loop? 2162 writeFrames = 0; 2163 2164 cacheParameters_l(); 2165 sleepTime = idleSleepTime; 2166 2167if (mType == MIXER) { 2168 sleepTimeShift = 0; 2169} 2170 2171 CpuStats cpuStats; 2172 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2173 2174 acquireWakeLock(); 2175 2176 while (!exitPending()) 2177 { 2178 cpuStats.sample(myName); 2179 2180 Vector< sp<EffectChain> > effectChains; 2181 2182 processConfigEvents(); 2183 2184 { // scope for mLock 2185 2186 Mutex::Autolock _l(mLock); 2187 2188 if (checkForNewParameters_l()) { 2189 cacheParameters_l(); 2190 } 2191 2192 saveOutputTracks(); 2193 2194 // put audio hardware into standby after short delay 2195 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2196 mSuspended > 0)) { 2197 if (!mStandby) { 2198 2199 threadLoop_standby(); 2200 2201 mStandby = true; 2202 mBytesWritten = 0; 2203 } 2204 2205 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2206 // we're about to wait, flush the binder command buffer 2207 IPCThreadState::self()->flushCommands(); 2208 2209 clearOutputTracks(); 2210 2211 if (exitPending()) break; 2212 2213 releaseWakeLock_l(); 2214 // wait until we have something to do... 2215 ALOGV("%s going to sleep", myName.string()); 2216 mWaitWorkCV.wait(mLock); 2217 ALOGV("%s waking up", myName.string()); 2218 acquireWakeLock_l(); 2219 2220 mPrevMixerStatus = MIXER_IDLE; 2221 2222 checkSilentMode_l(); 2223 2224 standbyTime = systemTime() + standbyDelay; 2225 sleepTime = idleSleepTime; 2226 if (mType == MIXER) { 2227 sleepTimeShift = 0; 2228 } 2229 2230 continue; 2231 } 2232 } 2233 2234 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2235 // Shift in the new status; this could be a queue if it's 2236 // useful to filter the mixer status over several cycles. 2237 mPrevMixerStatus = mMixerStatus; 2238 mMixerStatus = newMixerStatus; 2239 2240 // prevent any changes in effect chain list and in each effect chain 2241 // during mixing and effect process as the audio buffers could be deleted 2242 // or modified if an effect is created or deleted 2243 lockEffectChains_l(effectChains); 2244 } 2245 2246 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2247 threadLoop_mix(); 2248 } else { 2249 threadLoop_sleepTime(); 2250 } 2251 2252 if (mSuspended > 0) { 2253 sleepTime = suspendSleepTimeUs(); 2254 } 2255 2256 // only process effects if we're going to write 2257 if (sleepTime == 0) { 2258 for (size_t i = 0; i < effectChains.size(); i ++) { 2259 effectChains[i]->process_l(); 2260 } 2261 } 2262 2263 // enable changes in effect chain 2264 unlockEffectChains(effectChains); 2265 2266 // sleepTime == 0 means we must write to audio hardware 2267 if (sleepTime == 0) { 2268 2269 threadLoop_write(); 2270 2271if (mType == MIXER) { 2272 // write blocked detection 2273 nsecs_t now = systemTime(); 2274 nsecs_t delta = now - mLastWriteTime; 2275 if (!mStandby && delta > maxPeriod) { 2276 mNumDelayedWrites++; 2277 if ((now - lastWarning) > kWarningThrottleNs) { 2278 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2279 ns2ms(delta), mNumDelayedWrites, this); 2280 lastWarning = now; 2281 } 2282 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2283 // a different threshold. Or completely removed for what it is worth anyway... 2284 if (mStandby) { 2285 longStandbyExit = true; 2286 } 2287 } 2288} 2289 2290 mStandby = false; 2291 } else { 2292 usleep(sleepTime); 2293 } 2294 2295 // finally let go of removed track(s), without the lock held 2296 // since we can't guarantee the destructors won't acquire that 2297 // same lock. 2298 tracksToRemove.clear(); 2299 2300 // FIXME I don't understand the need for this here; 2301 // it was in the original code but maybe the 2302 // assignment in saveOutputTracks() makes this unnecessary? 2303 clearOutputTracks(); 2304 2305 // Effect chains will be actually deleted here if they were removed from 2306 // mEffectChains list during mixing or effects processing 2307 effectChains.clear(); 2308 2309 // FIXME Note that the above .clear() is no longer necessary since effectChains 2310 // is now local to this block, but will keep it for now (at least until merge done). 2311 } 2312 2313if (mType == MIXER || mType == DIRECT) { 2314 // put output stream into standby mode 2315 if (!mStandby) { 2316 mOutput->stream->common.standby(&mOutput->stream->common); 2317 } 2318} 2319if (mType == DUPLICATING) { 2320 // for DuplicatingThread, standby mode is handled by the outputTracks 2321} 2322 2323 releaseWakeLock(); 2324 2325 ALOGV("Thread %p type %d exiting", this, mType); 2326 return false; 2327} 2328 2329// shared by MIXER and DIRECT, overridden by DUPLICATING 2330void AudioFlinger::PlaybackThread::threadLoop_write() 2331{ 2332 // FIXME rewrite to reduce number of system calls 2333 mLastWriteTime = systemTime(); 2334 mInWrite = true; 2335 mBytesWritten += mixBufferSize; 2336 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2337 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2338 mNumWrites++; 2339 mInWrite = false; 2340} 2341 2342// shared by MIXER and DIRECT, overridden by DUPLICATING 2343void AudioFlinger::PlaybackThread::threadLoop_standby() 2344{ 2345 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2346 mOutput->stream->common.standby(&mOutput->stream->common); 2347} 2348 2349void AudioFlinger::MixerThread::threadLoop_mix() 2350{ 2351 // obtain the presentation timestamp of the next output buffer 2352 int64_t pts; 2353 status_t status = INVALID_OPERATION; 2354 2355 if (NULL != mOutput->stream->get_next_write_timestamp) { 2356 status = mOutput->stream->get_next_write_timestamp( 2357 mOutput->stream, &pts); 2358 } 2359 2360 if (status != NO_ERROR) { 2361 pts = AudioBufferProvider::kInvalidPTS; 2362 } 2363 2364 // mix buffers... 2365 mAudioMixer->process(pts); 2366 // increase sleep time progressively when application underrun condition clears. 2367 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2368 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2369 // such that we would underrun the audio HAL. 2370 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2371 sleepTimeShift--; 2372 } 2373 sleepTime = 0; 2374 standbyTime = systemTime() + standbyDelay; 2375 //TODO: delay standby when effects have a tail 2376} 2377 2378void AudioFlinger::MixerThread::threadLoop_sleepTime() 2379{ 2380 // If no tracks are ready, sleep once for the duration of an output 2381 // buffer size, then write 0s to the output 2382 if (sleepTime == 0) { 2383 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2384 sleepTime = activeSleepTime >> sleepTimeShift; 2385 if (sleepTime < kMinThreadSleepTimeUs) { 2386 sleepTime = kMinThreadSleepTimeUs; 2387 } 2388 // reduce sleep time in case of consecutive application underruns to avoid 2389 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2390 // duration we would end up writing less data than needed by the audio HAL if 2391 // the condition persists. 2392 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2393 sleepTimeShift++; 2394 } 2395 } else { 2396 sleepTime = idleSleepTime; 2397 } 2398 } else if (mBytesWritten != 0 || 2399 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2400 memset (mMixBuffer, 0, mixBufferSize); 2401 sleepTime = 0; 2402 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2403 } 2404 // TODO add standby time extension fct of effect tail 2405} 2406 2407// prepareTracks_l() must be called with ThreadBase::mLock held 2408AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2409 Vector< sp<Track> > *tracksToRemove) 2410{ 2411 2412 mixer_state mixerStatus = MIXER_IDLE; 2413 // find out which tracks need to be processed 2414 size_t count = mActiveTracks.size(); 2415 size_t mixedTracks = 0; 2416 size_t tracksWithEffect = 0; 2417 2418 float masterVolume = mMasterVolume; 2419 bool masterMute = mMasterMute; 2420 2421 if (masterMute) { 2422 masterVolume = 0; 2423 } 2424 // Delegate master volume control to effect in output mix effect chain if needed 2425 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2426 if (chain != 0) { 2427 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2428 chain->setVolume_l(&v, &v); 2429 masterVolume = (float)((v + (1 << 23)) >> 24); 2430 chain.clear(); 2431 } 2432 2433 for (size_t i=0 ; i<count ; i++) { 2434 sp<Track> t = mActiveTracks[i].promote(); 2435 if (t == 0) continue; 2436 2437 // this const just means the local variable doesn't change 2438 Track* const track = t.get(); 2439 audio_track_cblk_t* cblk = track->cblk(); 2440 2441 // The first time a track is added we wait 2442 // for all its buffers to be filled before processing it 2443 int name = track->name(); 2444 // make sure that we have enough frames to mix one full buffer. 2445 // enforce this condition only once to enable draining the buffer in case the client 2446 // app does not call stop() and relies on underrun to stop: 2447 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2448 // during last round 2449 uint32_t minFrames = 1; 2450 if (!track->isStopped() && !track->isPausing() && 2451 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2452 if (t->sampleRate() == (int)mSampleRate) { 2453 minFrames = mFrameCount; 2454 } else { 2455 // +1 for rounding and +1 for additional sample needed for interpolation 2456 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2457 // add frames already consumed but not yet released by the resampler 2458 // because cblk->framesReady() will include these frames 2459 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2460 // the minimum track buffer size is normally twice the number of frames necessary 2461 // to fill one buffer and the resampler should not leave more than one buffer worth 2462 // of unreleased frames after each pass, but just in case... 2463 ALOG_ASSERT(minFrames <= cblk->frameCount); 2464 } 2465 } 2466 if ((track->framesReady() >= minFrames) && track->isReady() && 2467 !track->isPaused() && !track->isTerminated()) 2468 { 2469 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2470 2471 mixedTracks++; 2472 2473 // track->mainBuffer() != mMixBuffer means there is an effect chain 2474 // connected to the track 2475 chain.clear(); 2476 if (track->mainBuffer() != mMixBuffer) { 2477 chain = getEffectChain_l(track->sessionId()); 2478 // Delegate volume control to effect in track effect chain if needed 2479 if (chain != 0) { 2480 tracksWithEffect++; 2481 } else { 2482 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2483 name, track->sessionId()); 2484 } 2485 } 2486 2487 2488 int param = AudioMixer::VOLUME; 2489 if (track->mFillingUpStatus == Track::FS_FILLED) { 2490 // no ramp for the first volume setting 2491 track->mFillingUpStatus = Track::FS_ACTIVE; 2492 if (track->mState == TrackBase::RESUMING) { 2493 track->mState = TrackBase::ACTIVE; 2494 param = AudioMixer::RAMP_VOLUME; 2495 } 2496 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2497 } else if (cblk->server != 0) { 2498 // If the track is stopped before the first frame was mixed, 2499 // do not apply ramp 2500 param = AudioMixer::RAMP_VOLUME; 2501 } 2502 2503 // compute volume for this track 2504 uint32_t vl, vr, va; 2505 if (track->isMuted() || track->isPausing() || 2506 mStreamTypes[track->streamType()].mute) { 2507 vl = vr = va = 0; 2508 if (track->isPausing()) { 2509 track->setPaused(); 2510 } 2511 } else { 2512 2513 // read original volumes with volume control 2514 float typeVolume = mStreamTypes[track->streamType()].volume; 2515 float v = masterVolume * typeVolume; 2516 uint32_t vlr = cblk->getVolumeLR(); 2517 vl = vlr & 0xFFFF; 2518 vr = vlr >> 16; 2519 // track volumes come from shared memory, so can't be trusted and must be clamped 2520 if (vl > MAX_GAIN_INT) { 2521 ALOGV("Track left volume out of range: %04X", vl); 2522 vl = MAX_GAIN_INT; 2523 } 2524 if (vr > MAX_GAIN_INT) { 2525 ALOGV("Track right volume out of range: %04X", vr); 2526 vr = MAX_GAIN_INT; 2527 } 2528 // now apply the master volume and stream type volume 2529 vl = (uint32_t)(v * vl) << 12; 2530 vr = (uint32_t)(v * vr) << 12; 2531 // assuming master volume and stream type volume each go up to 1.0, 2532 // vl and vr are now in 8.24 format 2533 2534 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2535 // send level comes from shared memory and so may be corrupt 2536 if (sendLevel > MAX_GAIN_INT) { 2537 ALOGV("Track send level out of range: %04X", sendLevel); 2538 sendLevel = MAX_GAIN_INT; 2539 } 2540 va = (uint32_t)(v * sendLevel); 2541 } 2542 // Delegate volume control to effect in track effect chain if needed 2543 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2544 // Do not ramp volume if volume is controlled by effect 2545 param = AudioMixer::VOLUME; 2546 track->mHasVolumeController = true; 2547 } else { 2548 // force no volume ramp when volume controller was just disabled or removed 2549 // from effect chain to avoid volume spike 2550 if (track->mHasVolumeController) { 2551 param = AudioMixer::VOLUME; 2552 } 2553 track->mHasVolumeController = false; 2554 } 2555 2556 // Convert volumes from 8.24 to 4.12 format 2557 // This additional clamping is needed in case chain->setVolume_l() overshot 2558 vl = (vl + (1 << 11)) >> 12; 2559 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2560 vr = (vr + (1 << 11)) >> 12; 2561 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2562 2563 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2564 2565 // XXX: these things DON'T need to be done each time 2566 mAudioMixer->setBufferProvider(name, track); 2567 mAudioMixer->enable(name); 2568 2569 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2570 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2571 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2572 mAudioMixer->setParameter( 2573 name, 2574 AudioMixer::TRACK, 2575 AudioMixer::FORMAT, (void *)track->format()); 2576 mAudioMixer->setParameter( 2577 name, 2578 AudioMixer::TRACK, 2579 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2580 mAudioMixer->setParameter( 2581 name, 2582 AudioMixer::RESAMPLE, 2583 AudioMixer::SAMPLE_RATE, 2584 (void *)(cblk->sampleRate)); 2585 mAudioMixer->setParameter( 2586 name, 2587 AudioMixer::TRACK, 2588 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2589 mAudioMixer->setParameter( 2590 name, 2591 AudioMixer::TRACK, 2592 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2593 2594 // reset retry count 2595 track->mRetryCount = kMaxTrackRetries; 2596 2597 // If one track is ready, set the mixer ready if: 2598 // - the mixer was not ready during previous round OR 2599 // - no other track is not ready 2600 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2601 mixerStatus != MIXER_TRACKS_ENABLED) { 2602 mixerStatus = MIXER_TRACKS_READY; 2603 } 2604 } else { 2605 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2606 if (track->isStopped()) { 2607 track->reset(); 2608 } 2609 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2610 // We have consumed all the buffers of this track. 2611 // Remove it from the list of active tracks. 2612 // TODO: use actual buffer filling status instead of latency when available from 2613 // audio HAL 2614 size_t audioHALFrames = 2615 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2616 size_t framesWritten = 2617 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2618 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2619 tracksToRemove->add(track); 2620 } 2621 } else { 2622 // No buffers for this track. Give it a few chances to 2623 // fill a buffer, then remove it from active list. 2624 if (--(track->mRetryCount) <= 0) { 2625 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2626 tracksToRemove->add(track); 2627 // indicate to client process that the track was disabled because of underrun 2628 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2629 // If one track is not ready, mark the mixer also not ready if: 2630 // - the mixer was ready during previous round OR 2631 // - no other track is ready 2632 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2633 mixerStatus != MIXER_TRACKS_READY) { 2634 mixerStatus = MIXER_TRACKS_ENABLED; 2635 } 2636 } 2637 mAudioMixer->disable(name); 2638 } 2639 } 2640 2641 // remove all the tracks that need to be... 2642 count = tracksToRemove->size(); 2643 if (CC_UNLIKELY(count)) { 2644 for (size_t i=0 ; i<count ; i++) { 2645 const sp<Track>& track = tracksToRemove->itemAt(i); 2646 mActiveTracks.remove(track); 2647 if (track->mainBuffer() != mMixBuffer) { 2648 chain = getEffectChain_l(track->sessionId()); 2649 if (chain != 0) { 2650 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2651 chain->decActiveTrackCnt(); 2652 } 2653 } 2654 if (track->isTerminated()) { 2655 removeTrack_l(track); 2656 } 2657 } 2658 } 2659 2660 // mix buffer must be cleared if all tracks are connected to an 2661 // effect chain as in this case the mixer will not write to 2662 // mix buffer and track effects will accumulate into it 2663 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2664 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2665 } 2666 2667 return mixerStatus; 2668} 2669 2670/* 2671The derived values that are cached: 2672 - mixBufferSize from frame count * frame size 2673 - activeSleepTime from activeSleepTimeUs() 2674 - idleSleepTime from idleSleepTimeUs() 2675 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2676 - maxPeriod from frame count and sample rate (MIXER only) 2677 2678The parameters that affect these derived values are: 2679 - frame count 2680 - frame size 2681 - sample rate 2682 - device type: A2DP or not 2683 - device latency 2684 - format: PCM or not 2685 - active sleep time 2686 - idle sleep time 2687*/ 2688 2689void AudioFlinger::PlaybackThread::cacheParameters_l() 2690{ 2691 mixBufferSize = mFrameCount * mFrameSize; 2692 activeSleepTime = activeSleepTimeUs(); 2693 idleSleepTime = idleSleepTimeUs(); 2694} 2695 2696void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2697{ 2698 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2699 this, streamType, mTracks.size()); 2700 Mutex::Autolock _l(mLock); 2701 2702 size_t size = mTracks.size(); 2703 for (size_t i = 0; i < size; i++) { 2704 sp<Track> t = mTracks[i]; 2705 if (t->streamType() == streamType) { 2706 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2707 t->mCblk->cv.signal(); 2708 } 2709 } 2710} 2711 2712void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2713{ 2714 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2715 this, streamType, valid); 2716 Mutex::Autolock _l(mLock); 2717 2718 mStreamTypes[streamType].valid = valid; 2719} 2720 2721// getTrackName_l() must be called with ThreadBase::mLock held 2722int AudioFlinger::MixerThread::getTrackName_l() 2723{ 2724 return mAudioMixer->getTrackName(); 2725} 2726 2727// deleteTrackName_l() must be called with ThreadBase::mLock held 2728void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2729{ 2730 ALOGV("remove track (%d) and delete from mixer", name); 2731 mAudioMixer->deleteTrackName(name); 2732} 2733 2734// checkForNewParameters_l() must be called with ThreadBase::mLock held 2735bool AudioFlinger::MixerThread::checkForNewParameters_l() 2736{ 2737 bool reconfig = false; 2738 2739 while (!mNewParameters.isEmpty()) { 2740 status_t status = NO_ERROR; 2741 String8 keyValuePair = mNewParameters[0]; 2742 AudioParameter param = AudioParameter(keyValuePair); 2743 int value; 2744 2745 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2746 reconfig = true; 2747 } 2748 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2749 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2750 status = BAD_VALUE; 2751 } else { 2752 reconfig = true; 2753 } 2754 } 2755 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2756 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2757 status = BAD_VALUE; 2758 } else { 2759 reconfig = true; 2760 } 2761 } 2762 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2763 // do not accept frame count changes if tracks are open as the track buffer 2764 // size depends on frame count and correct behavior would not be guaranteed 2765 // if frame count is changed after track creation 2766 if (!mTracks.isEmpty()) { 2767 status = INVALID_OPERATION; 2768 } else { 2769 reconfig = true; 2770 } 2771 } 2772 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2773#ifdef ADD_BATTERY_DATA 2774 // when changing the audio output device, call addBatteryData to notify 2775 // the change 2776 if ((int)mDevice != value) { 2777 uint32_t params = 0; 2778 // check whether speaker is on 2779 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2780 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2781 } 2782 2783 int deviceWithoutSpeaker 2784 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2785 // check if any other device (except speaker) is on 2786 if (value & deviceWithoutSpeaker ) { 2787 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2788 } 2789 2790 if (params != 0) { 2791 addBatteryData(params); 2792 } 2793 } 2794#endif 2795 2796 // forward device change to effects that have requested to be 2797 // aware of attached audio device. 2798 mDevice = (uint32_t)value; 2799 for (size_t i = 0; i < mEffectChains.size(); i++) { 2800 mEffectChains[i]->setDevice_l(mDevice); 2801 } 2802 } 2803 2804 if (status == NO_ERROR) { 2805 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2806 keyValuePair.string()); 2807 if (!mStandby && status == INVALID_OPERATION) { 2808 mOutput->stream->common.standby(&mOutput->stream->common); 2809 mStandby = true; 2810 mBytesWritten = 0; 2811 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2812 keyValuePair.string()); 2813 } 2814 if (status == NO_ERROR && reconfig) { 2815 delete mAudioMixer; 2816 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2817 mAudioMixer = NULL; 2818 readOutputParameters(); 2819 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2820 for (size_t i = 0; i < mTracks.size() ; i++) { 2821 int name = getTrackName_l(); 2822 if (name < 0) break; 2823 mTracks[i]->mName = name; 2824 // limit track sample rate to 2 x new output sample rate 2825 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2826 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2827 } 2828 } 2829 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2830 } 2831 } 2832 2833 mNewParameters.removeAt(0); 2834 2835 mParamStatus = status; 2836 mParamCond.signal(); 2837 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2838 // already timed out waiting for the status and will never signal the condition. 2839 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2840 } 2841 return reconfig; 2842} 2843 2844status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2845{ 2846 const size_t SIZE = 256; 2847 char buffer[SIZE]; 2848 String8 result; 2849 2850 PlaybackThread::dumpInternals(fd, args); 2851 2852 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2853 result.append(buffer); 2854 write(fd, result.string(), result.size()); 2855 return NO_ERROR; 2856} 2857 2858uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 2859{ 2860 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2861} 2862 2863uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 2864{ 2865 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2866} 2867 2868void AudioFlinger::MixerThread::cacheParameters_l() 2869{ 2870 PlaybackThread::cacheParameters_l(); 2871 2872 // FIXME: Relaxed timing because of a certain device that can't meet latency 2873 // Should be reduced to 2x after the vendor fixes the driver issue 2874 // increase threshold again due to low power audio mode. The way this warning 2875 // threshold is calculated and its usefulness should be reconsidered anyway. 2876 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2877} 2878 2879// ---------------------------------------------------------------------------- 2880AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2881 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2882 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2883 // mLeftVolFloat, mRightVolFloat 2884 // mLeftVolShort, mRightVolShort 2885{ 2886} 2887 2888AudioFlinger::DirectOutputThread::~DirectOutputThread() 2889{ 2890} 2891 2892AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2893 Vector< sp<Track> > *tracksToRemove 2894) 2895{ 2896 sp<Track> trackToRemove; 2897 2898 mixer_state mixerStatus = MIXER_IDLE; 2899 2900 // find out which tracks need to be processed 2901 if (mActiveTracks.size() != 0) { 2902 sp<Track> t = mActiveTracks[0].promote(); 2903 // The track died recently 2904 if (t == 0) return MIXER_IDLE; 2905 2906 Track* const track = t.get(); 2907 audio_track_cblk_t* cblk = track->cblk(); 2908 2909 // The first time a track is added we wait 2910 // for all its buffers to be filled before processing it 2911 if (cblk->framesReady() && track->isReady() && 2912 !track->isPaused() && !track->isTerminated()) 2913 { 2914 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2915 2916 if (track->mFillingUpStatus == Track::FS_FILLED) { 2917 track->mFillingUpStatus = Track::FS_ACTIVE; 2918 mLeftVolFloat = mRightVolFloat = 0; 2919 mLeftVolShort = mRightVolShort = 0; 2920 if (track->mState == TrackBase::RESUMING) { 2921 track->mState = TrackBase::ACTIVE; 2922 rampVolume = true; 2923 } 2924 } else if (cblk->server != 0) { 2925 // If the track is stopped before the first frame was mixed, 2926 // do not apply ramp 2927 rampVolume = true; 2928 } 2929 // compute volume for this track 2930 float left, right; 2931 if (track->isMuted() || mMasterMute || track->isPausing() || 2932 mStreamTypes[track->streamType()].mute) { 2933 left = right = 0; 2934 if (track->isPausing()) { 2935 track->setPaused(); 2936 } 2937 } else { 2938 float typeVolume = mStreamTypes[track->streamType()].volume; 2939 float v = mMasterVolume * typeVolume; 2940 uint32_t vlr = cblk->getVolumeLR(); 2941 float v_clamped = v * (vlr & 0xFFFF); 2942 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2943 left = v_clamped/MAX_GAIN; 2944 v_clamped = v * (vlr >> 16); 2945 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2946 right = v_clamped/MAX_GAIN; 2947 } 2948 2949 if (left != mLeftVolFloat || right != mRightVolFloat) { 2950 mLeftVolFloat = left; 2951 mRightVolFloat = right; 2952 2953 // If audio HAL implements volume control, 2954 // force software volume to nominal value 2955 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2956 left = 1.0f; 2957 right = 1.0f; 2958 } 2959 2960 // Convert volumes from float to 8.24 2961 uint32_t vl = (uint32_t)(left * (1 << 24)); 2962 uint32_t vr = (uint32_t)(right * (1 << 24)); 2963 2964 // Delegate volume control to effect in track effect chain if needed 2965 // only one effect chain can be present on DirectOutputThread, so if 2966 // there is one, the track is connected to it 2967 if (!mEffectChains.isEmpty()) { 2968 // Do not ramp volume if volume is controlled by effect 2969 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2970 rampVolume = false; 2971 } 2972 } 2973 2974 // Convert volumes from 8.24 to 4.12 format 2975 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2976 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2977 leftVol = (uint16_t)v_clamped; 2978 v_clamped = (vr + (1 << 11)) >> 12; 2979 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2980 rightVol = (uint16_t)v_clamped; 2981 } else { 2982 leftVol = mLeftVolShort; 2983 rightVol = mRightVolShort; 2984 rampVolume = false; 2985 } 2986 2987 // reset retry count 2988 track->mRetryCount = kMaxTrackRetriesDirect; 2989 mActiveTrack = t; 2990 mixerStatus = MIXER_TRACKS_READY; 2991 } else { 2992 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2993 if (track->isStopped()) { 2994 track->reset(); 2995 } 2996 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2997 // We have consumed all the buffers of this track. 2998 // Remove it from the list of active tracks. 2999 // TODO: implement behavior for compressed audio 3000 size_t audioHALFrames = 3001 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3002 size_t framesWritten = 3003 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3004 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3005 trackToRemove = track; 3006 } 3007 } else { 3008 // No buffers for this track. Give it a few chances to 3009 // fill a buffer, then remove it from active list. 3010 if (--(track->mRetryCount) <= 0) { 3011 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3012 trackToRemove = track; 3013 } else { 3014 mixerStatus = MIXER_TRACKS_ENABLED; 3015 } 3016 } 3017 } 3018 } 3019 3020 // FIXME merge this with similar code for removing multiple tracks 3021 // remove all the tracks that need to be... 3022 if (CC_UNLIKELY(trackToRemove != 0)) { 3023 tracksToRemove->add(trackToRemove); 3024 mActiveTracks.remove(trackToRemove); 3025 if (!mEffectChains.isEmpty()) { 3026 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3027 trackToRemove->sessionId()); 3028 mEffectChains[0]->decActiveTrackCnt(); 3029 } 3030 if (trackToRemove->isTerminated()) { 3031 removeTrack_l(trackToRemove); 3032 } 3033 } 3034 3035 return mixerStatus; 3036} 3037 3038void AudioFlinger::DirectOutputThread::threadLoop_mix() 3039{ 3040 AudioBufferProvider::Buffer buffer; 3041 size_t frameCount = mFrameCount; 3042 int8_t *curBuf = (int8_t *)mMixBuffer; 3043 // output audio to hardware 3044 while (frameCount) { 3045 buffer.frameCount = frameCount; 3046 mActiveTrack->getNextBuffer(&buffer); 3047 if (CC_UNLIKELY(buffer.raw == NULL)) { 3048 memset(curBuf, 0, frameCount * mFrameSize); 3049 break; 3050 } 3051 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3052 frameCount -= buffer.frameCount; 3053 curBuf += buffer.frameCount * mFrameSize; 3054 mActiveTrack->releaseBuffer(&buffer); 3055 } 3056 sleepTime = 0; 3057 standbyTime = systemTime() + standbyDelay; 3058 mActiveTrack.clear(); 3059 3060 // apply volume 3061 3062 // Do not apply volume on compressed audio 3063 if (!audio_is_linear_pcm(mFormat)) { 3064 return; 3065 } 3066 3067 // convert to signed 16 bit before volume calculation 3068 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3069 size_t count = mFrameCount * mChannelCount; 3070 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3071 int16_t *dst = mMixBuffer + count-1; 3072 while (count--) { 3073 *dst-- = (int16_t)(*src--^0x80) << 8; 3074 } 3075 } 3076 3077 frameCount = mFrameCount; 3078 int16_t *out = mMixBuffer; 3079 if (rampVolume) { 3080 if (mChannelCount == 1) { 3081 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3082 int32_t vlInc = d / (int32_t)frameCount; 3083 int32_t vl = ((int32_t)mLeftVolShort << 16); 3084 do { 3085 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3086 out++; 3087 vl += vlInc; 3088 } while (--frameCount); 3089 3090 } else { 3091 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3092 int32_t vlInc = d / (int32_t)frameCount; 3093 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3094 int32_t vrInc = d / (int32_t)frameCount; 3095 int32_t vl = ((int32_t)mLeftVolShort << 16); 3096 int32_t vr = ((int32_t)mRightVolShort << 16); 3097 do { 3098 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3099 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3100 out += 2; 3101 vl += vlInc; 3102 vr += vrInc; 3103 } while (--frameCount); 3104 } 3105 } else { 3106 if (mChannelCount == 1) { 3107 do { 3108 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3109 out++; 3110 } while (--frameCount); 3111 } else { 3112 do { 3113 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3114 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3115 out += 2; 3116 } while (--frameCount); 3117 } 3118 } 3119 3120 // convert back to unsigned 8 bit after volume calculation 3121 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3122 size_t count = mFrameCount * mChannelCount; 3123 int16_t *src = mMixBuffer; 3124 uint8_t *dst = (uint8_t *)mMixBuffer; 3125 while (count--) { 3126 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3127 } 3128 } 3129 3130 mLeftVolShort = leftVol; 3131 mRightVolShort = rightVol; 3132} 3133 3134void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3135{ 3136 if (sleepTime == 0) { 3137 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3138 sleepTime = activeSleepTime; 3139 } else { 3140 sleepTime = idleSleepTime; 3141 } 3142 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3143 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3144 sleepTime = 0; 3145 } 3146} 3147 3148// getTrackName_l() must be called with ThreadBase::mLock held 3149int AudioFlinger::DirectOutputThread::getTrackName_l() 3150{ 3151 return 0; 3152} 3153 3154// deleteTrackName_l() must be called with ThreadBase::mLock held 3155void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3156{ 3157} 3158 3159// checkForNewParameters_l() must be called with ThreadBase::mLock held 3160bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3161{ 3162 bool reconfig = false; 3163 3164 while (!mNewParameters.isEmpty()) { 3165 status_t status = NO_ERROR; 3166 String8 keyValuePair = mNewParameters[0]; 3167 AudioParameter param = AudioParameter(keyValuePair); 3168 int value; 3169 3170 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3171 // do not accept frame count changes if tracks are open as the track buffer 3172 // size depends on frame count and correct behavior would not be garantied 3173 // if frame count is changed after track creation 3174 if (!mTracks.isEmpty()) { 3175 status = INVALID_OPERATION; 3176 } else { 3177 reconfig = true; 3178 } 3179 } 3180 if (status == NO_ERROR) { 3181 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3182 keyValuePair.string()); 3183 if (!mStandby && status == INVALID_OPERATION) { 3184 mOutput->stream->common.standby(&mOutput->stream->common); 3185 mStandby = true; 3186 mBytesWritten = 0; 3187 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3188 keyValuePair.string()); 3189 } 3190 if (status == NO_ERROR && reconfig) { 3191 readOutputParameters(); 3192 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3193 } 3194 } 3195 3196 mNewParameters.removeAt(0); 3197 3198 mParamStatus = status; 3199 mParamCond.signal(); 3200 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3201 // already timed out waiting for the status and will never signal the condition. 3202 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3203 } 3204 return reconfig; 3205} 3206 3207uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3208{ 3209 uint32_t time; 3210 if (audio_is_linear_pcm(mFormat)) { 3211 time = PlaybackThread::activeSleepTimeUs(); 3212 } else { 3213 time = 10000; 3214 } 3215 return time; 3216} 3217 3218uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3219{ 3220 uint32_t time; 3221 if (audio_is_linear_pcm(mFormat)) { 3222 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3223 } else { 3224 time = 10000; 3225 } 3226 return time; 3227} 3228 3229uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3230{ 3231 uint32_t time; 3232 if (audio_is_linear_pcm(mFormat)) { 3233 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3234 } else { 3235 time = 10000; 3236 } 3237 return time; 3238} 3239 3240void AudioFlinger::DirectOutputThread::cacheParameters_l() 3241{ 3242 PlaybackThread::cacheParameters_l(); 3243 3244 // use shorter standby delay as on normal output to release 3245 // hardware resources as soon as possible 3246 standbyDelay = microseconds(activeSleepTime*2); 3247} 3248 3249// ---------------------------------------------------------------------------- 3250 3251AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3252 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3253 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3254 mWaitTimeMs(UINT_MAX) 3255{ 3256 addOutputTrack(mainThread); 3257} 3258 3259AudioFlinger::DuplicatingThread::~DuplicatingThread() 3260{ 3261 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3262 mOutputTracks[i]->destroy(); 3263 } 3264} 3265 3266void AudioFlinger::DuplicatingThread::threadLoop_mix() 3267{ 3268 // mix buffers... 3269 if (outputsReady(outputTracks)) { 3270 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3271 } else { 3272 memset(mMixBuffer, 0, mixBufferSize); 3273 } 3274 sleepTime = 0; 3275 writeFrames = mFrameCount; 3276} 3277 3278void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3279{ 3280 if (sleepTime == 0) { 3281 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3282 sleepTime = activeSleepTime; 3283 } else { 3284 sleepTime = idleSleepTime; 3285 } 3286 } else if (mBytesWritten != 0) { 3287 // flush remaining overflow buffers in output tracks 3288 for (size_t i = 0; i < outputTracks.size(); i++) { 3289 if (outputTracks[i]->isActive()) { 3290 sleepTime = 0; 3291 writeFrames = 0; 3292 memset(mMixBuffer, 0, mixBufferSize); 3293 break; 3294 } 3295 } 3296 } 3297} 3298 3299void AudioFlinger::DuplicatingThread::threadLoop_write() 3300{ 3301 standbyTime = systemTime() + standbyDelay; 3302 for (size_t i = 0; i < outputTracks.size(); i++) { 3303 outputTracks[i]->write(mMixBuffer, writeFrames); 3304 } 3305 mBytesWritten += mixBufferSize; 3306} 3307 3308void AudioFlinger::DuplicatingThread::threadLoop_standby() 3309{ 3310 // DuplicatingThread implements standby by stopping all tracks 3311 for (size_t i = 0; i < outputTracks.size(); i++) { 3312 outputTracks[i]->stop(); 3313 } 3314} 3315 3316void AudioFlinger::DuplicatingThread::saveOutputTracks() 3317{ 3318 outputTracks = mOutputTracks; 3319} 3320 3321void AudioFlinger::DuplicatingThread::clearOutputTracks() 3322{ 3323 outputTracks.clear(); 3324} 3325 3326void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3327{ 3328 Mutex::Autolock _l(mLock); 3329 // FIXME explain this formula 3330 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3331 OutputTrack *outputTrack = new OutputTrack(thread, 3332 this, 3333 mSampleRate, 3334 mFormat, 3335 mChannelMask, 3336 frameCount); 3337 if (outputTrack->cblk() != NULL) { 3338 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3339 mOutputTracks.add(outputTrack); 3340 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3341 updateWaitTime_l(); 3342 } 3343} 3344 3345void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3346{ 3347 Mutex::Autolock _l(mLock); 3348 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3349 if (mOutputTracks[i]->thread() == thread) { 3350 mOutputTracks[i]->destroy(); 3351 mOutputTracks.removeAt(i); 3352 updateWaitTime_l(); 3353 return; 3354 } 3355 } 3356 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3357} 3358 3359// caller must hold mLock 3360void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3361{ 3362 mWaitTimeMs = UINT_MAX; 3363 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3364 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3365 if (strong != 0) { 3366 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3367 if (waitTimeMs < mWaitTimeMs) { 3368 mWaitTimeMs = waitTimeMs; 3369 } 3370 } 3371 } 3372} 3373 3374 3375bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3376{ 3377 for (size_t i = 0; i < outputTracks.size(); i++) { 3378 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3379 if (thread == 0) { 3380 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3381 return false; 3382 } 3383 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3384 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3385 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3386 return false; 3387 } 3388 } 3389 return true; 3390} 3391 3392uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3393{ 3394 return (mWaitTimeMs * 1000) / 2; 3395} 3396 3397void AudioFlinger::DuplicatingThread::cacheParameters_l() 3398{ 3399 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3400 updateWaitTime_l(); 3401 3402 MixerThread::cacheParameters_l(); 3403} 3404 3405// ---------------------------------------------------------------------------- 3406 3407// TrackBase constructor must be called with AudioFlinger::mLock held 3408AudioFlinger::ThreadBase::TrackBase::TrackBase( 3409 ThreadBase *thread, 3410 const sp<Client>& client, 3411 uint32_t sampleRate, 3412 audio_format_t format, 3413 uint32_t channelMask, 3414 int frameCount, 3415 const sp<IMemory>& sharedBuffer, 3416 int sessionId) 3417 : RefBase(), 3418 mThread(thread), 3419 mClient(client), 3420 mCblk(NULL), 3421 // mBuffer 3422 // mBufferEnd 3423 mFrameCount(0), 3424 mState(IDLE), 3425 mFormat(format), 3426 mStepServerFailed(false), 3427 mSessionId(sessionId) 3428 // mChannelCount 3429 // mChannelMask 3430{ 3431 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3432 3433 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3434 size_t size = sizeof(audio_track_cblk_t); 3435 uint8_t channelCount = popcount(channelMask); 3436 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3437 if (sharedBuffer == 0) { 3438 size += bufferSize; 3439 } 3440 3441 if (client != NULL) { 3442 mCblkMemory = client->heap()->allocate(size); 3443 if (mCblkMemory != 0) { 3444 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3445 if (mCblk != NULL) { // construct the shared structure in-place. 3446 new(mCblk) audio_track_cblk_t(); 3447 // clear all buffers 3448 mCblk->frameCount = frameCount; 3449 mCblk->sampleRate = sampleRate; 3450// uncomment the following lines to quickly test 32-bit wraparound 3451// mCblk->user = 0xffff0000; 3452// mCblk->server = 0xffff0000; 3453// mCblk->userBase = 0xffff0000; 3454// mCblk->serverBase = 0xffff0000; 3455 mChannelCount = channelCount; 3456 mChannelMask = channelMask; 3457 if (sharedBuffer == 0) { 3458 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3459 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3460 // Force underrun condition to avoid false underrun callback until first data is 3461 // written to buffer (other flags are cleared) 3462 mCblk->flags = CBLK_UNDERRUN_ON; 3463 } else { 3464 mBuffer = sharedBuffer->pointer(); 3465 } 3466 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3467 } 3468 } else { 3469 ALOGE("not enough memory for AudioTrack size=%u", size); 3470 client->heap()->dump("AudioTrack"); 3471 return; 3472 } 3473 } else { 3474 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3475 // construct the shared structure in-place. 3476 new(mCblk) audio_track_cblk_t(); 3477 // clear all buffers 3478 mCblk->frameCount = frameCount; 3479 mCblk->sampleRate = sampleRate; 3480// uncomment the following lines to quickly test 32-bit wraparound 3481// mCblk->user = 0xffff0000; 3482// mCblk->server = 0xffff0000; 3483// mCblk->userBase = 0xffff0000; 3484// mCblk->serverBase = 0xffff0000; 3485 mChannelCount = channelCount; 3486 mChannelMask = channelMask; 3487 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3488 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3489 // Force underrun condition to avoid false underrun callback until first data is 3490 // written to buffer (other flags are cleared) 3491 mCblk->flags = CBLK_UNDERRUN_ON; 3492 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3493 } 3494} 3495 3496AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3497{ 3498 if (mCblk != NULL) { 3499 if (mClient == 0) { 3500 delete mCblk; 3501 } else { 3502 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3503 } 3504 } 3505 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3506 if (mClient != 0) { 3507 // Client destructor must run with AudioFlinger mutex locked 3508 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3509 // If the client's reference count drops to zero, the associated destructor 3510 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3511 // relying on the automatic clear() at end of scope. 3512 mClient.clear(); 3513 } 3514} 3515 3516// AudioBufferProvider interface 3517// getNextBuffer() = 0; 3518// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3519void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3520{ 3521 buffer->raw = NULL; 3522 mFrameCount = buffer->frameCount; 3523 (void) step(); // ignore return value of step() 3524 buffer->frameCount = 0; 3525} 3526 3527bool AudioFlinger::ThreadBase::TrackBase::step() { 3528 bool result; 3529 audio_track_cblk_t* cblk = this->cblk(); 3530 3531 result = cblk->stepServer(mFrameCount); 3532 if (!result) { 3533 ALOGV("stepServer failed acquiring cblk mutex"); 3534 mStepServerFailed = true; 3535 } 3536 return result; 3537} 3538 3539void AudioFlinger::ThreadBase::TrackBase::reset() { 3540 audio_track_cblk_t* cblk = this->cblk(); 3541 3542 cblk->user = 0; 3543 cblk->server = 0; 3544 cblk->userBase = 0; 3545 cblk->serverBase = 0; 3546 mStepServerFailed = false; 3547 ALOGV("TrackBase::reset"); 3548} 3549 3550int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3551 return (int)mCblk->sampleRate; 3552} 3553 3554void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3555 audio_track_cblk_t* cblk = this->cblk(); 3556 size_t frameSize = cblk->frameSize; 3557 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3558 int8_t *bufferEnd = bufferStart + frames * frameSize; 3559 3560 // Check validity of returned pointer in case the track control block would have been corrupted. 3561 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3562 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3563 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3564 server %u, serverBase %u, user %u, userBase %u", 3565 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3566 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3567 return NULL; 3568 } 3569 3570 return bufferStart; 3571} 3572 3573status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3574{ 3575 mSyncEvents.add(event); 3576 return NO_ERROR; 3577} 3578 3579// ---------------------------------------------------------------------------- 3580 3581// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3582AudioFlinger::PlaybackThread::Track::Track( 3583 PlaybackThread *thread, 3584 const sp<Client>& client, 3585 audio_stream_type_t streamType, 3586 uint32_t sampleRate, 3587 audio_format_t format, 3588 uint32_t channelMask, 3589 int frameCount, 3590 const sp<IMemory>& sharedBuffer, 3591 int sessionId, 3592 IAudioFlinger::track_flags_t flags) 3593 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3594 mMute(false), 3595 // mFillingUpStatus ? 3596 // mRetryCount initialized later when needed 3597 mSharedBuffer(sharedBuffer), 3598 mStreamType(streamType), 3599 mName(-1), // see note below 3600 mMainBuffer(thread->mixBuffer()), 3601 mAuxBuffer(NULL), 3602 mAuxEffectId(0), mHasVolumeController(false), 3603 mPresentationCompleteFrames(0), 3604 mFlags(flags) 3605{ 3606 if (mCblk != NULL) { 3607 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3608 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3609 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3610 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3611 mName = thread->getTrackName_l(); 3612 if (mName < 0) { 3613 ALOGE("no more track names available"); 3614 } 3615 } 3616 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3617} 3618 3619AudioFlinger::PlaybackThread::Track::~Track() 3620{ 3621 ALOGV("PlaybackThread::Track destructor"); 3622 sp<ThreadBase> thread = mThread.promote(); 3623 if (thread != 0) { 3624 Mutex::Autolock _l(thread->mLock); 3625 mState = TERMINATED; 3626 } 3627} 3628 3629void AudioFlinger::PlaybackThread::Track::destroy() 3630{ 3631 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3632 // by removing it from mTracks vector, so there is a risk that this Tracks's 3633 // destructor is called. As the destructor needs to lock mLock, 3634 // we must acquire a strong reference on this Track before locking mLock 3635 // here so that the destructor is called only when exiting this function. 3636 // On the other hand, as long as Track::destroy() is only called by 3637 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3638 // this Track with its member mTrack. 3639 sp<Track> keep(this); 3640 { // scope for mLock 3641 sp<ThreadBase> thread = mThread.promote(); 3642 if (thread != 0) { 3643 if (!isOutputTrack()) { 3644 if (mState == ACTIVE || mState == RESUMING) { 3645 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3646 3647#ifdef ADD_BATTERY_DATA 3648 // to track the speaker usage 3649 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3650#endif 3651 } 3652 AudioSystem::releaseOutput(thread->id()); 3653 } 3654 Mutex::Autolock _l(thread->mLock); 3655 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3656 playbackThread->destroyTrack_l(this); 3657 } 3658 } 3659} 3660 3661void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3662{ 3663 uint32_t vlr = mCblk->getVolumeLR(); 3664 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", 3665 mName - AudioMixer::TRACK0, 3666 (mClient == 0) ? getpid_cached : mClient->pid(), 3667 mStreamType, 3668 mFormat, 3669 mChannelMask, 3670 mSessionId, 3671 mFrameCount, 3672 mState, 3673 mMute, 3674 mFillingUpStatus, 3675 mCblk->sampleRate, 3676 vlr & 0xFFFF, 3677 vlr >> 16, 3678 mCblk->server, 3679 mCblk->user, 3680 (int)mMainBuffer, 3681 (int)mAuxBuffer); 3682} 3683 3684// AudioBufferProvider interface 3685status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3686 AudioBufferProvider::Buffer* buffer, int64_t pts) 3687{ 3688 audio_track_cblk_t* cblk = this->cblk(); 3689 uint32_t framesReady; 3690 uint32_t framesReq = buffer->frameCount; 3691 3692 // Check if last stepServer failed, try to step now 3693 if (mStepServerFailed) { 3694 if (!step()) goto getNextBuffer_exit; 3695 ALOGV("stepServer recovered"); 3696 mStepServerFailed = false; 3697 } 3698 3699 framesReady = cblk->framesReady(); 3700 3701 if (CC_LIKELY(framesReady)) { 3702 uint32_t s = cblk->server; 3703 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3704 3705 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3706 if (framesReq > framesReady) { 3707 framesReq = framesReady; 3708 } 3709 if (framesReq > bufferEnd - s) { 3710 framesReq = bufferEnd - s; 3711 } 3712 3713 buffer->raw = getBuffer(s, framesReq); 3714 if (buffer->raw == NULL) goto getNextBuffer_exit; 3715 3716 buffer->frameCount = framesReq; 3717 return NO_ERROR; 3718 } 3719 3720getNextBuffer_exit: 3721 buffer->raw = NULL; 3722 buffer->frameCount = 0; 3723 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3724 return NOT_ENOUGH_DATA; 3725} 3726 3727uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3728 return mCblk->framesReady(); 3729} 3730 3731bool AudioFlinger::PlaybackThread::Track::isReady() const { 3732 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3733 3734 if (framesReady() >= mCblk->frameCount || 3735 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3736 mFillingUpStatus = FS_FILLED; 3737 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3738 return true; 3739 } 3740 return false; 3741} 3742 3743status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid, 3744 AudioSystem::sync_event_t event, 3745 int triggerSession) 3746{ 3747 status_t status = NO_ERROR; 3748 ALOGV("start(%d), calling pid %d session %d tid %d", 3749 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3750 // check for use case 2 with missing callback 3751 if (isFastTrack() && (mSharedBuffer == 0) && (tid == 0)) { 3752 ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied"); 3753 mFlags &= ~IAudioFlinger::TRACK_FAST; 3754 // FIXME the track must be invalidated and moved to another thread or 3755 // attached directly to the normal mixer now 3756 } 3757 sp<ThreadBase> thread = mThread.promote(); 3758 if (thread != 0) { 3759 Mutex::Autolock _l(thread->mLock); 3760 track_state state = mState; 3761 // here the track could be either new, or restarted 3762 // in both cases "unstop" the track 3763 if (mState == PAUSED) { 3764 mState = TrackBase::RESUMING; 3765 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3766 } else { 3767 mState = TrackBase::ACTIVE; 3768 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3769 } 3770 3771 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3772 thread->mLock.unlock(); 3773 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3774 thread->mLock.lock(); 3775 3776#ifdef ADD_BATTERY_DATA 3777 // to track the speaker usage 3778 if (status == NO_ERROR) { 3779 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3780 } 3781#endif 3782 } 3783 if (status == NO_ERROR) { 3784 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3785 playbackThread->addTrack_l(this); 3786 } else { 3787 mState = state; 3788 } 3789 } else { 3790 status = BAD_VALUE; 3791 } 3792 return status; 3793} 3794 3795void AudioFlinger::PlaybackThread::Track::stop() 3796{ 3797 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3798 sp<ThreadBase> thread = mThread.promote(); 3799 if (thread != 0) { 3800 Mutex::Autolock _l(thread->mLock); 3801 track_state state = mState; 3802 if (mState > STOPPED) { 3803 mState = STOPPED; 3804 // If the track is not active (PAUSED and buffers full), flush buffers 3805 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3806 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3807 reset(); 3808 } 3809 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3810 } 3811 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3812 thread->mLock.unlock(); 3813 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3814 thread->mLock.lock(); 3815 3816#ifdef ADD_BATTERY_DATA 3817 // to track the speaker usage 3818 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3819#endif 3820 } 3821 } 3822} 3823 3824void AudioFlinger::PlaybackThread::Track::pause() 3825{ 3826 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3827 sp<ThreadBase> thread = mThread.promote(); 3828 if (thread != 0) { 3829 Mutex::Autolock _l(thread->mLock); 3830 if (mState == ACTIVE || mState == RESUMING) { 3831 mState = PAUSING; 3832 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3833 if (!isOutputTrack()) { 3834 thread->mLock.unlock(); 3835 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3836 thread->mLock.lock(); 3837 3838#ifdef ADD_BATTERY_DATA 3839 // to track the speaker usage 3840 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3841#endif 3842 } 3843 } 3844 } 3845} 3846 3847void AudioFlinger::PlaybackThread::Track::flush() 3848{ 3849 ALOGV("flush(%d)", mName); 3850 sp<ThreadBase> thread = mThread.promote(); 3851 if (thread != 0) { 3852 Mutex::Autolock _l(thread->mLock); 3853 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3854 return; 3855 } 3856 // No point remaining in PAUSED state after a flush => go to 3857 // STOPPED state 3858 mState = STOPPED; 3859 3860 // do not reset the track if it is still in the process of being stopped or paused. 3861 // this will be done by prepareTracks_l() when the track is stopped. 3862 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3863 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3864 reset(); 3865 } 3866 } 3867} 3868 3869void AudioFlinger::PlaybackThread::Track::reset() 3870{ 3871 // Do not reset twice to avoid discarding data written just after a flush and before 3872 // the audioflinger thread detects the track is stopped. 3873 if (!mResetDone) { 3874 TrackBase::reset(); 3875 // Force underrun condition to avoid false underrun callback until first data is 3876 // written to buffer 3877 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3878 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3879 mFillingUpStatus = FS_FILLING; 3880 mResetDone = true; 3881 mPresentationCompleteFrames = 0; 3882 } 3883} 3884 3885void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3886{ 3887 mMute = muted; 3888} 3889 3890status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3891{ 3892 status_t status = DEAD_OBJECT; 3893 sp<ThreadBase> thread = mThread.promote(); 3894 if (thread != 0) { 3895 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3896 status = playbackThread->attachAuxEffect(this, EffectId); 3897 } 3898 return status; 3899} 3900 3901void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3902{ 3903 mAuxEffectId = EffectId; 3904 mAuxBuffer = buffer; 3905} 3906 3907bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 3908 size_t audioHalFrames) 3909{ 3910 // a track is considered presented when the total number of frames written to audio HAL 3911 // corresponds to the number of frames written when presentationComplete() is called for the 3912 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 3913 if (mPresentationCompleteFrames == 0) { 3914 mPresentationCompleteFrames = framesWritten + audioHalFrames; 3915 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 3916 mPresentationCompleteFrames, audioHalFrames); 3917 } 3918 if (framesWritten >= mPresentationCompleteFrames) { 3919 ALOGV("presentationComplete() session %d complete: framesWritten %d", 3920 mSessionId, framesWritten); 3921 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 3922 mPresentationCompleteFrames = 0; 3923 return true; 3924 } 3925 return false; 3926} 3927 3928void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 3929{ 3930 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 3931 if (mSyncEvents[i]->type() == type) { 3932 mSyncEvents[i]->trigger(); 3933 mSyncEvents.removeAt(i); 3934 i--; 3935 } 3936 } 3937} 3938 3939 3940// timed audio tracks 3941 3942sp<AudioFlinger::PlaybackThread::TimedTrack> 3943AudioFlinger::PlaybackThread::TimedTrack::create( 3944 PlaybackThread *thread, 3945 const sp<Client>& client, 3946 audio_stream_type_t streamType, 3947 uint32_t sampleRate, 3948 audio_format_t format, 3949 uint32_t channelMask, 3950 int frameCount, 3951 const sp<IMemory>& sharedBuffer, 3952 int sessionId) { 3953 if (!client->reserveTimedTrack()) 3954 return NULL; 3955 3956 return new TimedTrack( 3957 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3958 sharedBuffer, sessionId); 3959} 3960 3961AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3962 PlaybackThread *thread, 3963 const sp<Client>& client, 3964 audio_stream_type_t streamType, 3965 uint32_t sampleRate, 3966 audio_format_t format, 3967 uint32_t channelMask, 3968 int frameCount, 3969 const sp<IMemory>& sharedBuffer, 3970 int sessionId) 3971 : Track(thread, client, streamType, sampleRate, format, channelMask, 3972 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 3973 mTimedSilenceBuffer(NULL), 3974 mTimedSilenceBufferSize(0), 3975 mTimedAudioOutputOnTime(false), 3976 mMediaTimeTransformValid(false) 3977{ 3978 LocalClock lc; 3979 mLocalTimeFreq = lc.getLocalFreq(); 3980 3981 mLocalTimeToSampleTransform.a_zero = 0; 3982 mLocalTimeToSampleTransform.b_zero = 0; 3983 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3984 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3985 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3986 &mLocalTimeToSampleTransform.a_to_b_denom); 3987} 3988 3989AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3990 mClient->releaseTimedTrack(); 3991 delete [] mTimedSilenceBuffer; 3992} 3993 3994status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3995 size_t size, sp<IMemory>* buffer) { 3996 3997 Mutex::Autolock _l(mTimedBufferQueueLock); 3998 3999 trimTimedBufferQueue_l(); 4000 4001 // lazily initialize the shared memory heap for timed buffers 4002 if (mTimedMemoryDealer == NULL) { 4003 const int kTimedBufferHeapSize = 512 << 10; 4004 4005 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4006 "AudioFlingerTimed"); 4007 if (mTimedMemoryDealer == NULL) 4008 return NO_MEMORY; 4009 } 4010 4011 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4012 if (newBuffer == NULL) { 4013 newBuffer = mTimedMemoryDealer->allocate(size); 4014 if (newBuffer == NULL) 4015 return NO_MEMORY; 4016 } 4017 4018 *buffer = newBuffer; 4019 return NO_ERROR; 4020} 4021 4022// caller must hold mTimedBufferQueueLock 4023void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4024 int64_t mediaTimeNow; 4025 { 4026 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4027 if (!mMediaTimeTransformValid) 4028 return; 4029 4030 int64_t targetTimeNow; 4031 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4032 ? mCCHelper.getCommonTime(&targetTimeNow) 4033 : mCCHelper.getLocalTime(&targetTimeNow); 4034 4035 if (OK != res) 4036 return; 4037 4038 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4039 &mediaTimeNow)) { 4040 return; 4041 } 4042 } 4043 4044 size_t trimIndex; 4045 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 4046 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 4047 break; 4048 } 4049 4050 if (trimIndex) { 4051 mTimedBufferQueue.removeItemsAt(0, trimIndex); 4052 } 4053} 4054 4055status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4056 const sp<IMemory>& buffer, int64_t pts) { 4057 4058 { 4059 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4060 if (!mMediaTimeTransformValid) 4061 return INVALID_OPERATION; 4062 } 4063 4064 Mutex::Autolock _l(mTimedBufferQueueLock); 4065 4066 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4067 4068 return NO_ERROR; 4069} 4070 4071status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4072 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4073 4074 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 4075 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4076 target); 4077 4078 if (!(target == TimedAudioTrack::LOCAL_TIME || 4079 target == TimedAudioTrack::COMMON_TIME)) { 4080 return BAD_VALUE; 4081 } 4082 4083 Mutex::Autolock lock(mMediaTimeTransformLock); 4084 mMediaTimeTransform = xform; 4085 mMediaTimeTransformTarget = target; 4086 mMediaTimeTransformValid = true; 4087 4088 return NO_ERROR; 4089} 4090 4091#define min(a, b) ((a) < (b) ? (a) : (b)) 4092 4093// implementation of getNextBuffer for tracks whose buffers have timestamps 4094status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4095 AudioBufferProvider::Buffer* buffer, int64_t pts) 4096{ 4097 if (pts == AudioBufferProvider::kInvalidPTS) { 4098 buffer->raw = 0; 4099 buffer->frameCount = 0; 4100 return INVALID_OPERATION; 4101 } 4102 4103 Mutex::Autolock _l(mTimedBufferQueueLock); 4104 4105 while (true) { 4106 4107 // if we have no timed buffers, then fail 4108 if (mTimedBufferQueue.isEmpty()) { 4109 buffer->raw = 0; 4110 buffer->frameCount = 0; 4111 return NOT_ENOUGH_DATA; 4112 } 4113 4114 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4115 4116 // calculate the PTS of the head of the timed buffer queue expressed in 4117 // local time 4118 int64_t headLocalPTS; 4119 { 4120 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4121 4122 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4123 4124 if (mMediaTimeTransform.a_to_b_denom == 0) { 4125 // the transform represents a pause, so yield silence 4126 timedYieldSilence(buffer->frameCount, buffer); 4127 return NO_ERROR; 4128 } 4129 4130 int64_t transformedPTS; 4131 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4132 &transformedPTS)) { 4133 // the transform failed. this shouldn't happen, but if it does 4134 // then just drop this buffer 4135 ALOGW("timedGetNextBuffer transform failed"); 4136 buffer->raw = 0; 4137 buffer->frameCount = 0; 4138 mTimedBufferQueue.removeAt(0); 4139 return NO_ERROR; 4140 } 4141 4142 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4143 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4144 &headLocalPTS)) { 4145 buffer->raw = 0; 4146 buffer->frameCount = 0; 4147 return INVALID_OPERATION; 4148 } 4149 } else { 4150 headLocalPTS = transformedPTS; 4151 } 4152 } 4153 4154 // adjust the head buffer's PTS to reflect the portion of the head buffer 4155 // that has already been consumed 4156 int64_t effectivePTS = headLocalPTS + 4157 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4158 4159 // Calculate the delta in samples between the head of the input buffer 4160 // queue and the start of the next output buffer that will be written. 4161 // If the transformation fails because of over or underflow, it means 4162 // that the sample's position in the output stream is so far out of 4163 // whack that it should just be dropped. 4164 int64_t sampleDelta; 4165 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4166 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4167 mTimedBufferQueue.removeAt(0); 4168 continue; 4169 } 4170 if (!mLocalTimeToSampleTransform.doForwardTransform( 4171 (effectivePTS - pts) << 32, &sampleDelta)) { 4172 ALOGV("*** too late during sample rate transform: dropped buffer"); 4173 mTimedBufferQueue.removeAt(0); 4174 continue; 4175 } 4176 4177 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4178 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4179 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4180 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4181 4182 // if the delta between the ideal placement for the next input sample and 4183 // the current output position is within this threshold, then we will 4184 // concatenate the next input samples to the previous output 4185 const int64_t kSampleContinuityThreshold = 4186 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4187 4188 // if this is the first buffer of audio that we're emitting from this track 4189 // then it should be almost exactly on time. 4190 const int64_t kSampleStartupThreshold = 1LL << 32; 4191 4192 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4193 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4194 // the next input is close enough to being on time, so concatenate it 4195 // with the last output 4196 timedYieldSamples(buffer); 4197 4198 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4199 return NO_ERROR; 4200 } else if (sampleDelta > 0) { 4201 // the gap between the current output position and the proper start of 4202 // the next input sample is too big, so fill it with silence 4203 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4204 4205 timedYieldSilence(framesUntilNextInput, buffer); 4206 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4207 return NO_ERROR; 4208 } else { 4209 // the next input sample is late 4210 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4211 size_t onTimeSamplePosition = 4212 head.position() + lateFrames * mCblk->frameSize; 4213 4214 if (onTimeSamplePosition > head.buffer()->size()) { 4215 // all the remaining samples in the head are too late, so 4216 // drop it and move on 4217 ALOGV("*** too late: dropped buffer"); 4218 mTimedBufferQueue.removeAt(0); 4219 continue; 4220 } else { 4221 // skip over the late samples 4222 head.setPosition(onTimeSamplePosition); 4223 4224 // yield the available samples 4225 timedYieldSamples(buffer); 4226 4227 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4228 return NO_ERROR; 4229 } 4230 } 4231 } 4232} 4233 4234// Yield samples from the timed buffer queue head up to the given output 4235// buffer's capacity. 4236// 4237// Caller must hold mTimedBufferQueueLock 4238void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4239 AudioBufferProvider::Buffer* buffer) { 4240 4241 const TimedBuffer& head = mTimedBufferQueue[0]; 4242 4243 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4244 head.position()); 4245 4246 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4247 mCblk->frameSize); 4248 size_t framesRequested = buffer->frameCount; 4249 buffer->frameCount = min(framesLeftInHead, framesRequested); 4250 4251 mTimedAudioOutputOnTime = true; 4252} 4253 4254// Yield samples of silence up to the given output buffer's capacity 4255// 4256// Caller must hold mTimedBufferQueueLock 4257void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4258 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4259 4260 // lazily allocate a buffer filled with silence 4261 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4262 delete [] mTimedSilenceBuffer; 4263 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4264 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4265 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4266 } 4267 4268 buffer->raw = mTimedSilenceBuffer; 4269 size_t framesRequested = buffer->frameCount; 4270 buffer->frameCount = min(numFrames, framesRequested); 4271 4272 mTimedAudioOutputOnTime = false; 4273} 4274 4275// AudioBufferProvider interface 4276void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4277 AudioBufferProvider::Buffer* buffer) { 4278 4279 Mutex::Autolock _l(mTimedBufferQueueLock); 4280 4281 // If the buffer which was just released is part of the buffer at the head 4282 // of the queue, be sure to update the amt of the buffer which has been 4283 // consumed. If the buffer being returned is not part of the head of the 4284 // queue, its either because the buffer is part of the silence buffer, or 4285 // because the head of the timed queue was trimmed after the mixer called 4286 // getNextBuffer but before the mixer called releaseBuffer. 4287 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4288 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4289 4290 void* start = head.buffer()->pointer(); 4291 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4292 4293 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4294 head.setPosition(head.position() + 4295 (buffer->frameCount * mCblk->frameSize)); 4296 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4297 mTimedBufferQueue.removeAt(0); 4298 } 4299 } 4300 } 4301 4302 buffer->raw = 0; 4303 buffer->frameCount = 0; 4304} 4305 4306uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4307 Mutex::Autolock _l(mTimedBufferQueueLock); 4308 4309 uint32_t frames = 0; 4310 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4311 const TimedBuffer& tb = mTimedBufferQueue[i]; 4312 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4313 } 4314 4315 return frames; 4316} 4317 4318AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4319 : mPTS(0), mPosition(0) {} 4320 4321AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4322 const sp<IMemory>& buffer, int64_t pts) 4323 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4324 4325// ---------------------------------------------------------------------------- 4326 4327// RecordTrack constructor must be called with AudioFlinger::mLock held 4328AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4329 RecordThread *thread, 4330 const sp<Client>& client, 4331 uint32_t sampleRate, 4332 audio_format_t format, 4333 uint32_t channelMask, 4334 int frameCount, 4335 int sessionId) 4336 : TrackBase(thread, client, sampleRate, format, 4337 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4338 mOverflow(false) 4339{ 4340 if (mCblk != NULL) { 4341 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4342 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4343 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4344 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4345 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4346 } else { 4347 mCblk->frameSize = sizeof(int8_t); 4348 } 4349 } 4350} 4351 4352AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4353{ 4354 sp<ThreadBase> thread = mThread.promote(); 4355 if (thread != 0) { 4356 AudioSystem::releaseInput(thread->id()); 4357 } 4358} 4359 4360// AudioBufferProvider interface 4361status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4362{ 4363 audio_track_cblk_t* cblk = this->cblk(); 4364 uint32_t framesAvail; 4365 uint32_t framesReq = buffer->frameCount; 4366 4367 // Check if last stepServer failed, try to step now 4368 if (mStepServerFailed) { 4369 if (!step()) goto getNextBuffer_exit; 4370 ALOGV("stepServer recovered"); 4371 mStepServerFailed = false; 4372 } 4373 4374 framesAvail = cblk->framesAvailable_l(); 4375 4376 if (CC_LIKELY(framesAvail)) { 4377 uint32_t s = cblk->server; 4378 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4379 4380 if (framesReq > framesAvail) { 4381 framesReq = framesAvail; 4382 } 4383 if (framesReq > bufferEnd - s) { 4384 framesReq = bufferEnd - s; 4385 } 4386 4387 buffer->raw = getBuffer(s, framesReq); 4388 if (buffer->raw == NULL) goto getNextBuffer_exit; 4389 4390 buffer->frameCount = framesReq; 4391 return NO_ERROR; 4392 } 4393 4394getNextBuffer_exit: 4395 buffer->raw = NULL; 4396 buffer->frameCount = 0; 4397 return NOT_ENOUGH_DATA; 4398} 4399 4400status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid, 4401 AudioSystem::sync_event_t event, 4402 int triggerSession) 4403{ 4404 sp<ThreadBase> thread = mThread.promote(); 4405 if (thread != 0) { 4406 RecordThread *recordThread = (RecordThread *)thread.get(); 4407 return recordThread->start(this, tid, event, triggerSession); 4408 } else { 4409 return BAD_VALUE; 4410 } 4411} 4412 4413void AudioFlinger::RecordThread::RecordTrack::stop() 4414{ 4415 sp<ThreadBase> thread = mThread.promote(); 4416 if (thread != 0) { 4417 RecordThread *recordThread = (RecordThread *)thread.get(); 4418 recordThread->stop(this); 4419 TrackBase::reset(); 4420 // Force overrun condition to avoid false overrun callback until first data is 4421 // read from buffer 4422 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4423 } 4424} 4425 4426void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4427{ 4428 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4429 (mClient == 0) ? getpid_cached : mClient->pid(), 4430 mFormat, 4431 mChannelMask, 4432 mSessionId, 4433 mFrameCount, 4434 mState, 4435 mCblk->sampleRate, 4436 mCblk->server, 4437 mCblk->user); 4438} 4439 4440 4441// ---------------------------------------------------------------------------- 4442 4443AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4444 PlaybackThread *playbackThread, 4445 DuplicatingThread *sourceThread, 4446 uint32_t sampleRate, 4447 audio_format_t format, 4448 uint32_t channelMask, 4449 int frameCount) 4450 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 4451 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 4452 mActive(false), mSourceThread(sourceThread) 4453{ 4454 4455 if (mCblk != NULL) { 4456 mCblk->flags |= CBLK_DIRECTION_OUT; 4457 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4458 mOutBuffer.frameCount = 0; 4459 playbackThread->mTracks.add(this); 4460 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4461 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4462 mCblk, mBuffer, mCblk->buffers, 4463 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4464 } else { 4465 ALOGW("Error creating output track on thread %p", playbackThread); 4466 } 4467} 4468 4469AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4470{ 4471 clearBufferQueue(); 4472} 4473 4474status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid, 4475 AudioSystem::sync_event_t event, 4476 int triggerSession) 4477{ 4478 status_t status = Track::start(tid, event, triggerSession); 4479 if (status != NO_ERROR) { 4480 return status; 4481 } 4482 4483 mActive = true; 4484 mRetryCount = 127; 4485 return status; 4486} 4487 4488void AudioFlinger::PlaybackThread::OutputTrack::stop() 4489{ 4490 Track::stop(); 4491 clearBufferQueue(); 4492 mOutBuffer.frameCount = 0; 4493 mActive = false; 4494} 4495 4496bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4497{ 4498 Buffer *pInBuffer; 4499 Buffer inBuffer; 4500 uint32_t channelCount = mChannelCount; 4501 bool outputBufferFull = false; 4502 inBuffer.frameCount = frames; 4503 inBuffer.i16 = data; 4504 4505 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4506 4507 if (!mActive && frames != 0) { 4508 start(0); 4509 sp<ThreadBase> thread = mThread.promote(); 4510 if (thread != 0) { 4511 MixerThread *mixerThread = (MixerThread *)thread.get(); 4512 if (mCblk->frameCount > frames){ 4513 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4514 uint32_t startFrames = (mCblk->frameCount - frames); 4515 pInBuffer = new Buffer; 4516 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4517 pInBuffer->frameCount = startFrames; 4518 pInBuffer->i16 = pInBuffer->mBuffer; 4519 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4520 mBufferQueue.add(pInBuffer); 4521 } else { 4522 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4523 } 4524 } 4525 } 4526 } 4527 4528 while (waitTimeLeftMs) { 4529 // First write pending buffers, then new data 4530 if (mBufferQueue.size()) { 4531 pInBuffer = mBufferQueue.itemAt(0); 4532 } else { 4533 pInBuffer = &inBuffer; 4534 } 4535 4536 if (pInBuffer->frameCount == 0) { 4537 break; 4538 } 4539 4540 if (mOutBuffer.frameCount == 0) { 4541 mOutBuffer.frameCount = pInBuffer->frameCount; 4542 nsecs_t startTime = systemTime(); 4543 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4544 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4545 outputBufferFull = true; 4546 break; 4547 } 4548 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4549 if (waitTimeLeftMs >= waitTimeMs) { 4550 waitTimeLeftMs -= waitTimeMs; 4551 } else { 4552 waitTimeLeftMs = 0; 4553 } 4554 } 4555 4556 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4557 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4558 mCblk->stepUser(outFrames); 4559 pInBuffer->frameCount -= outFrames; 4560 pInBuffer->i16 += outFrames * channelCount; 4561 mOutBuffer.frameCount -= outFrames; 4562 mOutBuffer.i16 += outFrames * channelCount; 4563 4564 if (pInBuffer->frameCount == 0) { 4565 if (mBufferQueue.size()) { 4566 mBufferQueue.removeAt(0); 4567 delete [] pInBuffer->mBuffer; 4568 delete pInBuffer; 4569 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4570 } else { 4571 break; 4572 } 4573 } 4574 } 4575 4576 // If we could not write all frames, allocate a buffer and queue it for next time. 4577 if (inBuffer.frameCount) { 4578 sp<ThreadBase> thread = mThread.promote(); 4579 if (thread != 0 && !thread->standby()) { 4580 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4581 pInBuffer = new Buffer; 4582 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4583 pInBuffer->frameCount = inBuffer.frameCount; 4584 pInBuffer->i16 = pInBuffer->mBuffer; 4585 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4586 mBufferQueue.add(pInBuffer); 4587 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4588 } else { 4589 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4590 } 4591 } 4592 } 4593 4594 // Calling write() with a 0 length buffer, means that no more data will be written: 4595 // If no more buffers are pending, fill output track buffer to make sure it is started 4596 // by output mixer. 4597 if (frames == 0 && mBufferQueue.size() == 0) { 4598 if (mCblk->user < mCblk->frameCount) { 4599 frames = mCblk->frameCount - mCblk->user; 4600 pInBuffer = new Buffer; 4601 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4602 pInBuffer->frameCount = frames; 4603 pInBuffer->i16 = pInBuffer->mBuffer; 4604 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4605 mBufferQueue.add(pInBuffer); 4606 } else if (mActive) { 4607 stop(); 4608 } 4609 } 4610 4611 return outputBufferFull; 4612} 4613 4614status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4615{ 4616 int active; 4617 status_t result; 4618 audio_track_cblk_t* cblk = mCblk; 4619 uint32_t framesReq = buffer->frameCount; 4620 4621// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4622 buffer->frameCount = 0; 4623 4624 uint32_t framesAvail = cblk->framesAvailable(); 4625 4626 4627 if (framesAvail == 0) { 4628 Mutex::Autolock _l(cblk->lock); 4629 goto start_loop_here; 4630 while (framesAvail == 0) { 4631 active = mActive; 4632 if (CC_UNLIKELY(!active)) { 4633 ALOGV("Not active and NO_MORE_BUFFERS"); 4634 return NO_MORE_BUFFERS; 4635 } 4636 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4637 if (result != NO_ERROR) { 4638 return NO_MORE_BUFFERS; 4639 } 4640 // read the server count again 4641 start_loop_here: 4642 framesAvail = cblk->framesAvailable_l(); 4643 } 4644 } 4645 4646// if (framesAvail < framesReq) { 4647// return NO_MORE_BUFFERS; 4648// } 4649 4650 if (framesReq > framesAvail) { 4651 framesReq = framesAvail; 4652 } 4653 4654 uint32_t u = cblk->user; 4655 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4656 4657 if (framesReq > bufferEnd - u) { 4658 framesReq = bufferEnd - u; 4659 } 4660 4661 buffer->frameCount = framesReq; 4662 buffer->raw = (void *)cblk->buffer(u); 4663 return NO_ERROR; 4664} 4665 4666 4667void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4668{ 4669 size_t size = mBufferQueue.size(); 4670 4671 for (size_t i = 0; i < size; i++) { 4672 Buffer *pBuffer = mBufferQueue.itemAt(i); 4673 delete [] pBuffer->mBuffer; 4674 delete pBuffer; 4675 } 4676 mBufferQueue.clear(); 4677} 4678 4679// ---------------------------------------------------------------------------- 4680 4681AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4682 : RefBase(), 4683 mAudioFlinger(audioFlinger), 4684 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4685 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4686 mPid(pid), 4687 mTimedTrackCount(0) 4688{ 4689 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4690} 4691 4692// Client destructor must be called with AudioFlinger::mLock held 4693AudioFlinger::Client::~Client() 4694{ 4695 mAudioFlinger->removeClient_l(mPid); 4696} 4697 4698sp<MemoryDealer> AudioFlinger::Client::heap() const 4699{ 4700 return mMemoryDealer; 4701} 4702 4703// Reserve one of the limited slots for a timed audio track associated 4704// with this client 4705bool AudioFlinger::Client::reserveTimedTrack() 4706{ 4707 const int kMaxTimedTracksPerClient = 4; 4708 4709 Mutex::Autolock _l(mTimedTrackLock); 4710 4711 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4712 ALOGW("can not create timed track - pid %d has exceeded the limit", 4713 mPid); 4714 return false; 4715 } 4716 4717 mTimedTrackCount++; 4718 return true; 4719} 4720 4721// Release a slot for a timed audio track 4722void AudioFlinger::Client::releaseTimedTrack() 4723{ 4724 Mutex::Autolock _l(mTimedTrackLock); 4725 mTimedTrackCount--; 4726} 4727 4728// ---------------------------------------------------------------------------- 4729 4730AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4731 const sp<IAudioFlingerClient>& client, 4732 pid_t pid) 4733 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4734{ 4735} 4736 4737AudioFlinger::NotificationClient::~NotificationClient() 4738{ 4739} 4740 4741void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4742{ 4743 sp<NotificationClient> keep(this); 4744 mAudioFlinger->removeNotificationClient(mPid); 4745} 4746 4747// ---------------------------------------------------------------------------- 4748 4749AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4750 : BnAudioTrack(), 4751 mTrack(track) 4752{ 4753} 4754 4755AudioFlinger::TrackHandle::~TrackHandle() { 4756 // just stop the track on deletion, associated resources 4757 // will be freed from the main thread once all pending buffers have 4758 // been played. Unless it's not in the active track list, in which 4759 // case we free everything now... 4760 mTrack->destroy(); 4761} 4762 4763sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4764 return mTrack->getCblk(); 4765} 4766 4767status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4768 return mTrack->start(tid); 4769} 4770 4771void AudioFlinger::TrackHandle::stop() { 4772 mTrack->stop(); 4773} 4774 4775void AudioFlinger::TrackHandle::flush() { 4776 mTrack->flush(); 4777} 4778 4779void AudioFlinger::TrackHandle::mute(bool e) { 4780 mTrack->mute(e); 4781} 4782 4783void AudioFlinger::TrackHandle::pause() { 4784 mTrack->pause(); 4785} 4786 4787status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4788{ 4789 return mTrack->attachAuxEffect(EffectId); 4790} 4791 4792status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4793 sp<IMemory>* buffer) { 4794 if (!mTrack->isTimedTrack()) 4795 return INVALID_OPERATION; 4796 4797 PlaybackThread::TimedTrack* tt = 4798 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4799 return tt->allocateTimedBuffer(size, buffer); 4800} 4801 4802status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4803 int64_t pts) { 4804 if (!mTrack->isTimedTrack()) 4805 return INVALID_OPERATION; 4806 4807 PlaybackThread::TimedTrack* tt = 4808 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4809 return tt->queueTimedBuffer(buffer, pts); 4810} 4811 4812status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4813 const LinearTransform& xform, int target) { 4814 4815 if (!mTrack->isTimedTrack()) 4816 return INVALID_OPERATION; 4817 4818 PlaybackThread::TimedTrack* tt = 4819 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4820 return tt->setMediaTimeTransform( 4821 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4822} 4823 4824status_t AudioFlinger::TrackHandle::onTransact( 4825 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4826{ 4827 return BnAudioTrack::onTransact(code, data, reply, flags); 4828} 4829 4830// ---------------------------------------------------------------------------- 4831 4832sp<IAudioRecord> AudioFlinger::openRecord( 4833 pid_t pid, 4834 audio_io_handle_t input, 4835 uint32_t sampleRate, 4836 audio_format_t format, 4837 uint32_t channelMask, 4838 int frameCount, 4839 IAudioFlinger::track_flags_t flags, 4840 int *sessionId, 4841 status_t *status) 4842{ 4843 sp<RecordThread::RecordTrack> recordTrack; 4844 sp<RecordHandle> recordHandle; 4845 sp<Client> client; 4846 status_t lStatus; 4847 RecordThread *thread; 4848 size_t inFrameCount; 4849 int lSessionId; 4850 4851 // check calling permissions 4852 if (!recordingAllowed()) { 4853 lStatus = PERMISSION_DENIED; 4854 goto Exit; 4855 } 4856 4857 // add client to list 4858 { // scope for mLock 4859 Mutex::Autolock _l(mLock); 4860 thread = checkRecordThread_l(input); 4861 if (thread == NULL) { 4862 lStatus = BAD_VALUE; 4863 goto Exit; 4864 } 4865 4866 client = registerPid_l(pid); 4867 4868 // If no audio session id is provided, create one here 4869 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4870 lSessionId = *sessionId; 4871 } else { 4872 lSessionId = nextUniqueId(); 4873 if (sessionId != NULL) { 4874 *sessionId = lSessionId; 4875 } 4876 } 4877 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4878 recordTrack = thread->createRecordTrack_l(client, 4879 sampleRate, 4880 format, 4881 channelMask, 4882 frameCount, 4883 lSessionId, 4884 &lStatus); 4885 } 4886 if (lStatus != NO_ERROR) { 4887 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4888 // destructor is called by the TrackBase destructor with mLock held 4889 client.clear(); 4890 recordTrack.clear(); 4891 goto Exit; 4892 } 4893 4894 // return to handle to client 4895 recordHandle = new RecordHandle(recordTrack); 4896 lStatus = NO_ERROR; 4897 4898Exit: 4899 if (status) { 4900 *status = lStatus; 4901 } 4902 return recordHandle; 4903} 4904 4905// ---------------------------------------------------------------------------- 4906 4907AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4908 : BnAudioRecord(), 4909 mRecordTrack(recordTrack) 4910{ 4911} 4912 4913AudioFlinger::RecordHandle::~RecordHandle() { 4914 stop(); 4915} 4916 4917sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4918 return mRecordTrack->getCblk(); 4919} 4920 4921status_t AudioFlinger::RecordHandle::start(pid_t tid, int event, int triggerSession) { 4922 ALOGV("RecordHandle::start()"); 4923 return mRecordTrack->start(tid, (AudioSystem::sync_event_t)event, triggerSession); 4924} 4925 4926void AudioFlinger::RecordHandle::stop() { 4927 ALOGV("RecordHandle::stop()"); 4928 mRecordTrack->stop(); 4929} 4930 4931status_t AudioFlinger::RecordHandle::onTransact( 4932 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4933{ 4934 return BnAudioRecord::onTransact(code, data, reply, flags); 4935} 4936 4937// ---------------------------------------------------------------------------- 4938 4939AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4940 AudioStreamIn *input, 4941 uint32_t sampleRate, 4942 uint32_t channels, 4943 audio_io_handle_t id, 4944 uint32_t device) : 4945 ThreadBase(audioFlinger, id, device, RECORD), 4946 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4947 // mRsmpInIndex and mInputBytes set by readInputParameters() 4948 mReqChannelCount(popcount(channels)), 4949 mReqSampleRate(sampleRate) 4950 // mBytesRead is only meaningful while active, and so is cleared in start() 4951 // (but might be better to also clear here for dump?) 4952{ 4953 snprintf(mName, kNameLength, "AudioIn_%X", id); 4954 4955 readInputParameters(); 4956} 4957 4958 4959AudioFlinger::RecordThread::~RecordThread() 4960{ 4961 delete[] mRsmpInBuffer; 4962 delete mResampler; 4963 delete[] mRsmpOutBuffer; 4964} 4965 4966void AudioFlinger::RecordThread::onFirstRef() 4967{ 4968 run(mName, PRIORITY_URGENT_AUDIO); 4969} 4970 4971status_t AudioFlinger::RecordThread::readyToRun() 4972{ 4973 status_t status = initCheck(); 4974 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4975 return status; 4976} 4977 4978bool AudioFlinger::RecordThread::threadLoop() 4979{ 4980 AudioBufferProvider::Buffer buffer; 4981 sp<RecordTrack> activeTrack; 4982 Vector< sp<EffectChain> > effectChains; 4983 4984 nsecs_t lastWarning = 0; 4985 4986 acquireWakeLock(); 4987 4988 // start recording 4989 while (!exitPending()) { 4990 4991 processConfigEvents(); 4992 4993 { // scope for mLock 4994 Mutex::Autolock _l(mLock); 4995 checkForNewParameters_l(); 4996 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4997 if (!mStandby) { 4998 mInput->stream->common.standby(&mInput->stream->common); 4999 mStandby = true; 5000 } 5001 5002 if (exitPending()) break; 5003 5004 releaseWakeLock_l(); 5005 ALOGV("RecordThread: loop stopping"); 5006 // go to sleep 5007 mWaitWorkCV.wait(mLock); 5008 ALOGV("RecordThread: loop starting"); 5009 acquireWakeLock_l(); 5010 continue; 5011 } 5012 if (mActiveTrack != 0) { 5013 if (mActiveTrack->mState == TrackBase::PAUSING) { 5014 if (!mStandby) { 5015 mInput->stream->common.standby(&mInput->stream->common); 5016 mStandby = true; 5017 } 5018 mActiveTrack.clear(); 5019 mStartStopCond.broadcast(); 5020 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5021 if (mReqChannelCount != mActiveTrack->channelCount()) { 5022 mActiveTrack.clear(); 5023 mStartStopCond.broadcast(); 5024 } else if (mBytesRead != 0) { 5025 // record start succeeds only if first read from audio input 5026 // succeeds 5027 if (mBytesRead > 0) { 5028 mActiveTrack->mState = TrackBase::ACTIVE; 5029 } else { 5030 mActiveTrack.clear(); 5031 } 5032 mStartStopCond.broadcast(); 5033 } 5034 mStandby = false; 5035 } 5036 } 5037 lockEffectChains_l(effectChains); 5038 } 5039 5040 if (mActiveTrack != 0) { 5041 if (mActiveTrack->mState != TrackBase::ACTIVE && 5042 mActiveTrack->mState != TrackBase::RESUMING) { 5043 unlockEffectChains(effectChains); 5044 usleep(kRecordThreadSleepUs); 5045 continue; 5046 } 5047 for (size_t i = 0; i < effectChains.size(); i ++) { 5048 effectChains[i]->process_l(); 5049 } 5050 5051 buffer.frameCount = mFrameCount; 5052 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5053 size_t framesOut = buffer.frameCount; 5054 if (mResampler == NULL) { 5055 // no resampling 5056 while (framesOut) { 5057 size_t framesIn = mFrameCount - mRsmpInIndex; 5058 if (framesIn) { 5059 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5060 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5061 if (framesIn > framesOut) 5062 framesIn = framesOut; 5063 mRsmpInIndex += framesIn; 5064 framesOut -= framesIn; 5065 if ((int)mChannelCount == mReqChannelCount || 5066 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5067 memcpy(dst, src, framesIn * mFrameSize); 5068 } else { 5069 int16_t *src16 = (int16_t *)src; 5070 int16_t *dst16 = (int16_t *)dst; 5071 if (mChannelCount == 1) { 5072 while (framesIn--) { 5073 *dst16++ = *src16; 5074 *dst16++ = *src16++; 5075 } 5076 } else { 5077 while (framesIn--) { 5078 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5079 src16 += 2; 5080 } 5081 } 5082 } 5083 } 5084 if (framesOut && mFrameCount == mRsmpInIndex) { 5085 if (framesOut == mFrameCount && 5086 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5087 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5088 framesOut = 0; 5089 } else { 5090 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5091 mRsmpInIndex = 0; 5092 } 5093 if (mBytesRead < 0) { 5094 ALOGE("Error reading audio input"); 5095 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5096 // Force input into standby so that it tries to 5097 // recover at next read attempt 5098 mInput->stream->common.standby(&mInput->stream->common); 5099 usleep(kRecordThreadSleepUs); 5100 } 5101 mRsmpInIndex = mFrameCount; 5102 framesOut = 0; 5103 buffer.frameCount = 0; 5104 } 5105 } 5106 } 5107 } else { 5108 // resampling 5109 5110 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5111 // alter output frame count as if we were expecting stereo samples 5112 if (mChannelCount == 1 && mReqChannelCount == 1) { 5113 framesOut >>= 1; 5114 } 5115 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5116 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5117 // are 32 bit aligned which should be always true. 5118 if (mChannelCount == 2 && mReqChannelCount == 1) { 5119 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5120 // the resampler always outputs stereo samples: do post stereo to mono conversion 5121 int16_t *src = (int16_t *)mRsmpOutBuffer; 5122 int16_t *dst = buffer.i16; 5123 while (framesOut--) { 5124 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5125 src += 2; 5126 } 5127 } else { 5128 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5129 } 5130 5131 } 5132 if (mFramestoDrop == 0) { 5133 mActiveTrack->releaseBuffer(&buffer); 5134 } else { 5135 if (mFramestoDrop > 0) { 5136 mFramestoDrop -= buffer.frameCount; 5137 if (mFramestoDrop < 0) { 5138 mFramestoDrop = 0; 5139 } 5140 } 5141 } 5142 mActiveTrack->overflow(); 5143 } 5144 // client isn't retrieving buffers fast enough 5145 else { 5146 if (!mActiveTrack->setOverflow()) { 5147 nsecs_t now = systemTime(); 5148 if ((now - lastWarning) > kWarningThrottleNs) { 5149 ALOGW("RecordThread: buffer overflow"); 5150 lastWarning = now; 5151 } 5152 } 5153 // Release the processor for a while before asking for a new buffer. 5154 // This will give the application more chance to read from the buffer and 5155 // clear the overflow. 5156 usleep(kRecordThreadSleepUs); 5157 } 5158 } 5159 // enable changes in effect chain 5160 unlockEffectChains(effectChains); 5161 effectChains.clear(); 5162 } 5163 5164 if (!mStandby) { 5165 mInput->stream->common.standby(&mInput->stream->common); 5166 } 5167 mActiveTrack.clear(); 5168 5169 mStartStopCond.broadcast(); 5170 5171 releaseWakeLock(); 5172 5173 ALOGV("RecordThread %p exiting", this); 5174 return false; 5175} 5176 5177 5178sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5179 const sp<AudioFlinger::Client>& client, 5180 uint32_t sampleRate, 5181 audio_format_t format, 5182 int channelMask, 5183 int frameCount, 5184 int sessionId, 5185 status_t *status) 5186{ 5187 sp<RecordTrack> track; 5188 status_t lStatus; 5189 5190 lStatus = initCheck(); 5191 if (lStatus != NO_ERROR) { 5192 ALOGE("Audio driver not initialized."); 5193 goto Exit; 5194 } 5195 5196 { // scope for mLock 5197 Mutex::Autolock _l(mLock); 5198 5199 track = new RecordTrack(this, client, sampleRate, 5200 format, channelMask, frameCount, sessionId); 5201 5202 if (track->getCblk() == 0) { 5203 lStatus = NO_MEMORY; 5204 goto Exit; 5205 } 5206 5207 mTrack = track.get(); 5208 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5209 bool suspend = audio_is_bluetooth_sco_device( 5210 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5211 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5212 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5213 } 5214 lStatus = NO_ERROR; 5215 5216Exit: 5217 if (status) { 5218 *status = lStatus; 5219 } 5220 return track; 5221} 5222 5223status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5224 pid_t tid, AudioSystem::sync_event_t event, 5225 int triggerSession) 5226{ 5227 ALOGV("RecordThread::start tid=%d, event %d, triggerSession %d", tid, event, triggerSession); 5228 sp<ThreadBase> strongMe = this; 5229 status_t status = NO_ERROR; 5230 5231 if (event == AudioSystem::SYNC_EVENT_NONE) { 5232 mSyncStartEvent.clear(); 5233 mFramestoDrop = 0; 5234 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5235 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5236 triggerSession, 5237 recordTrack->sessionId(), 5238 syncStartEventCallback, 5239 this); 5240 mFramestoDrop = -1; 5241 } 5242 5243 { 5244 AutoMutex lock(mLock); 5245 if (mActiveTrack != 0) { 5246 if (recordTrack != mActiveTrack.get()) { 5247 status = -EBUSY; 5248 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5249 mActiveTrack->mState = TrackBase::ACTIVE; 5250 } 5251 return status; 5252 } 5253 5254 recordTrack->mState = TrackBase::IDLE; 5255 mActiveTrack = recordTrack; 5256 mLock.unlock(); 5257 status_t status = AudioSystem::startInput(mId); 5258 mLock.lock(); 5259 if (status != NO_ERROR) { 5260 mActiveTrack.clear(); 5261 clearSyncStartEvent(); 5262 return status; 5263 } 5264 mRsmpInIndex = mFrameCount; 5265 mBytesRead = 0; 5266 if (mResampler != NULL) { 5267 mResampler->reset(); 5268 } 5269 mActiveTrack->mState = TrackBase::RESUMING; 5270 // signal thread to start 5271 ALOGV("Signal record thread"); 5272 mWaitWorkCV.signal(); 5273 // do not wait for mStartStopCond if exiting 5274 if (exitPending()) { 5275 mActiveTrack.clear(); 5276 status = INVALID_OPERATION; 5277 goto startError; 5278 } 5279 mStartStopCond.wait(mLock); 5280 if (mActiveTrack == 0) { 5281 ALOGV("Record failed to start"); 5282 status = BAD_VALUE; 5283 goto startError; 5284 } 5285 ALOGV("Record started OK"); 5286 return status; 5287 } 5288startError: 5289 AudioSystem::stopInput(mId); 5290 clearSyncStartEvent(); 5291 return status; 5292} 5293 5294void AudioFlinger::RecordThread::clearSyncStartEvent() 5295{ 5296 if (mSyncStartEvent != 0) { 5297 mSyncStartEvent->cancel(); 5298 } 5299 mSyncStartEvent.clear(); 5300} 5301 5302void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5303{ 5304 sp<SyncEvent> strongEvent = event.promote(); 5305 5306 if (strongEvent != 0) { 5307 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5308 me->handleSyncStartEvent(strongEvent); 5309 } 5310} 5311 5312void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5313{ 5314 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5315 mActiveTrack.get(), 5316 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5317 event->listenerSession()); 5318 5319 if (mActiveTrack != 0 && 5320 event == mSyncStartEvent) { 5321 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5322 // from audio HAL 5323 mFramestoDrop = mFrameCount * 2; 5324 mSyncStartEvent.clear(); 5325 } 5326} 5327 5328void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5329 ALOGV("RecordThread::stop"); 5330 sp<ThreadBase> strongMe = this; 5331 { 5332 AutoMutex lock(mLock); 5333 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5334 mActiveTrack->mState = TrackBase::PAUSING; 5335 // do not wait for mStartStopCond if exiting 5336 if (exitPending()) { 5337 return; 5338 } 5339 mStartStopCond.wait(mLock); 5340 // if we have been restarted, recordTrack == mActiveTrack.get() here 5341 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5342 mLock.unlock(); 5343 AudioSystem::stopInput(mId); 5344 mLock.lock(); 5345 ALOGV("Record stopped OK"); 5346 } 5347 } 5348 } 5349} 5350 5351bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5352{ 5353 return false; 5354} 5355 5356status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5357{ 5358 if (!isValidSyncEvent(event)) { 5359 return BAD_VALUE; 5360 } 5361 5362 Mutex::Autolock _l(mLock); 5363 5364 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5365 mTrack->setSyncEvent(event); 5366 return NO_ERROR; 5367 } 5368 return NAME_NOT_FOUND; 5369} 5370 5371status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5372{ 5373 const size_t SIZE = 256; 5374 char buffer[SIZE]; 5375 String8 result; 5376 5377 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5378 result.append(buffer); 5379 5380 if (mActiveTrack != 0) { 5381 result.append("Active Track:\n"); 5382 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5383 mActiveTrack->dump(buffer, SIZE); 5384 result.append(buffer); 5385 5386 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5387 result.append(buffer); 5388 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5389 result.append(buffer); 5390 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5391 result.append(buffer); 5392 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5393 result.append(buffer); 5394 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5395 result.append(buffer); 5396 5397 5398 } else { 5399 result.append("No record client\n"); 5400 } 5401 write(fd, result.string(), result.size()); 5402 5403 dumpBase(fd, args); 5404 dumpEffectChains(fd, args); 5405 5406 return NO_ERROR; 5407} 5408 5409// AudioBufferProvider interface 5410status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5411{ 5412 size_t framesReq = buffer->frameCount; 5413 size_t framesReady = mFrameCount - mRsmpInIndex; 5414 int channelCount; 5415 5416 if (framesReady == 0) { 5417 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5418 if (mBytesRead < 0) { 5419 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5420 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5421 // Force input into standby so that it tries to 5422 // recover at next read attempt 5423 mInput->stream->common.standby(&mInput->stream->common); 5424 usleep(kRecordThreadSleepUs); 5425 } 5426 buffer->raw = NULL; 5427 buffer->frameCount = 0; 5428 return NOT_ENOUGH_DATA; 5429 } 5430 mRsmpInIndex = 0; 5431 framesReady = mFrameCount; 5432 } 5433 5434 if (framesReq > framesReady) { 5435 framesReq = framesReady; 5436 } 5437 5438 if (mChannelCount == 1 && mReqChannelCount == 2) { 5439 channelCount = 1; 5440 } else { 5441 channelCount = 2; 5442 } 5443 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5444 buffer->frameCount = framesReq; 5445 return NO_ERROR; 5446} 5447 5448// AudioBufferProvider interface 5449void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5450{ 5451 mRsmpInIndex += buffer->frameCount; 5452 buffer->frameCount = 0; 5453} 5454 5455bool AudioFlinger::RecordThread::checkForNewParameters_l() 5456{ 5457 bool reconfig = false; 5458 5459 while (!mNewParameters.isEmpty()) { 5460 status_t status = NO_ERROR; 5461 String8 keyValuePair = mNewParameters[0]; 5462 AudioParameter param = AudioParameter(keyValuePair); 5463 int value; 5464 audio_format_t reqFormat = mFormat; 5465 int reqSamplingRate = mReqSampleRate; 5466 int reqChannelCount = mReqChannelCount; 5467 5468 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5469 reqSamplingRate = value; 5470 reconfig = true; 5471 } 5472 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5473 reqFormat = (audio_format_t) value; 5474 reconfig = true; 5475 } 5476 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5477 reqChannelCount = popcount(value); 5478 reconfig = true; 5479 } 5480 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5481 // do not accept frame count changes if tracks are open as the track buffer 5482 // size depends on frame count and correct behavior would not be guaranteed 5483 // if frame count is changed after track creation 5484 if (mActiveTrack != 0) { 5485 status = INVALID_OPERATION; 5486 } else { 5487 reconfig = true; 5488 } 5489 } 5490 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5491 // forward device change to effects that have requested to be 5492 // aware of attached audio device. 5493 for (size_t i = 0; i < mEffectChains.size(); i++) { 5494 mEffectChains[i]->setDevice_l(value); 5495 } 5496 // store input device and output device but do not forward output device to audio HAL. 5497 // Note that status is ignored by the caller for output device 5498 // (see AudioFlinger::setParameters() 5499 if (value & AUDIO_DEVICE_OUT_ALL) { 5500 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5501 status = BAD_VALUE; 5502 } else { 5503 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5504 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5505 if (mTrack != NULL) { 5506 bool suspend = audio_is_bluetooth_sco_device( 5507 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5508 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5509 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5510 } 5511 } 5512 mDevice |= (uint32_t)value; 5513 } 5514 if (status == NO_ERROR) { 5515 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5516 if (status == INVALID_OPERATION) { 5517 mInput->stream->common.standby(&mInput->stream->common); 5518 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5519 keyValuePair.string()); 5520 } 5521 if (reconfig) { 5522 if (status == BAD_VALUE && 5523 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5524 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5525 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5526 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5527 (reqChannelCount <= FCC_2)) { 5528 status = NO_ERROR; 5529 } 5530 if (status == NO_ERROR) { 5531 readInputParameters(); 5532 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5533 } 5534 } 5535 } 5536 5537 mNewParameters.removeAt(0); 5538 5539 mParamStatus = status; 5540 mParamCond.signal(); 5541 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5542 // already timed out waiting for the status and will never signal the condition. 5543 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5544 } 5545 return reconfig; 5546} 5547 5548String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5549{ 5550 char *s; 5551 String8 out_s8 = String8(); 5552 5553 Mutex::Autolock _l(mLock); 5554 if (initCheck() != NO_ERROR) { 5555 return out_s8; 5556 } 5557 5558 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5559 out_s8 = String8(s); 5560 free(s); 5561 return out_s8; 5562} 5563 5564void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5565 AudioSystem::OutputDescriptor desc; 5566 void *param2 = NULL; 5567 5568 switch (event) { 5569 case AudioSystem::INPUT_OPENED: 5570 case AudioSystem::INPUT_CONFIG_CHANGED: 5571 desc.channels = mChannelMask; 5572 desc.samplingRate = mSampleRate; 5573 desc.format = mFormat; 5574 desc.frameCount = mFrameCount; 5575 desc.latency = 0; 5576 param2 = &desc; 5577 break; 5578 5579 case AudioSystem::INPUT_CLOSED: 5580 default: 5581 break; 5582 } 5583 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5584} 5585 5586void AudioFlinger::RecordThread::readInputParameters() 5587{ 5588 delete mRsmpInBuffer; 5589 // mRsmpInBuffer is always assigned a new[] below 5590 delete mRsmpOutBuffer; 5591 mRsmpOutBuffer = NULL; 5592 delete mResampler; 5593 mResampler = NULL; 5594 5595 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5596 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5597 mChannelCount = (uint16_t)popcount(mChannelMask); 5598 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5599 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5600 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5601 mFrameCount = mInputBytes / mFrameSize; 5602 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5603 5604 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5605 { 5606 int channelCount; 5607 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5608 // stereo to mono post process as the resampler always outputs stereo. 5609 if (mChannelCount == 1 && mReqChannelCount == 2) { 5610 channelCount = 1; 5611 } else { 5612 channelCount = 2; 5613 } 5614 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5615 mResampler->setSampleRate(mSampleRate); 5616 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5617 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5618 5619 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5620 if (mChannelCount == 1 && mReqChannelCount == 1) { 5621 mFrameCount >>= 1; 5622 } 5623 5624 } 5625 mRsmpInIndex = mFrameCount; 5626} 5627 5628unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5629{ 5630 Mutex::Autolock _l(mLock); 5631 if (initCheck() != NO_ERROR) { 5632 return 0; 5633 } 5634 5635 return mInput->stream->get_input_frames_lost(mInput->stream); 5636} 5637 5638uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5639{ 5640 Mutex::Autolock _l(mLock); 5641 uint32_t result = 0; 5642 if (getEffectChain_l(sessionId) != 0) { 5643 result = EFFECT_SESSION; 5644 } 5645 5646 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5647 result |= TRACK_SESSION; 5648 } 5649 5650 return result; 5651} 5652 5653AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5654{ 5655 Mutex::Autolock _l(mLock); 5656 return mTrack; 5657} 5658 5659AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5660{ 5661 Mutex::Autolock _l(mLock); 5662 return mInput; 5663} 5664 5665AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5666{ 5667 Mutex::Autolock _l(mLock); 5668 AudioStreamIn *input = mInput; 5669 mInput = NULL; 5670 return input; 5671} 5672 5673// this method must always be called either with ThreadBase mLock held or inside the thread loop 5674audio_stream_t* AudioFlinger::RecordThread::stream() const 5675{ 5676 if (mInput == NULL) { 5677 return NULL; 5678 } 5679 return &mInput->stream->common; 5680} 5681 5682 5683// ---------------------------------------------------------------------------- 5684 5685audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5686 uint32_t *pSamplingRate, 5687 audio_format_t *pFormat, 5688 uint32_t *pChannels, 5689 uint32_t *pLatencyMs, 5690 audio_policy_output_flags_t flags) 5691{ 5692 status_t status; 5693 PlaybackThread *thread = NULL; 5694 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5695 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5696 uint32_t channels = pChannels ? *pChannels : 0; 5697 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5698 audio_stream_out_t *outStream; 5699 audio_hw_device_t *outHwDev; 5700 5701 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5702 pDevices ? *pDevices : 0, 5703 samplingRate, 5704 format, 5705 channels, 5706 flags); 5707 5708 if (pDevices == NULL || *pDevices == 0) { 5709 return 0; 5710 } 5711 5712 Mutex::Autolock _l(mLock); 5713 5714 outHwDev = findSuitableHwDev_l(*pDevices); 5715 if (outHwDev == NULL) 5716 return 0; 5717 5718 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5719 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5720 &channels, &samplingRate, &outStream); 5721 mHardwareStatus = AUDIO_HW_IDLE; 5722 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5723 outStream, 5724 samplingRate, 5725 format, 5726 channels, 5727 status); 5728 5729 if (outStream != NULL) { 5730 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5731 audio_io_handle_t id = nextUniqueId(); 5732 5733 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5734 (format != AUDIO_FORMAT_PCM_16_BIT) || 5735 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5736 thread = new DirectOutputThread(this, output, id, *pDevices); 5737 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5738 } else { 5739 thread = new MixerThread(this, output, id, *pDevices); 5740 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5741 } 5742 mPlaybackThreads.add(id, thread); 5743 5744 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5745 if (pFormat != NULL) *pFormat = format; 5746 if (pChannels != NULL) *pChannels = channels; 5747 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5748 5749 // notify client processes of the new output creation 5750 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5751 return id; 5752 } 5753 5754 return 0; 5755} 5756 5757audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5758 audio_io_handle_t output2) 5759{ 5760 Mutex::Autolock _l(mLock); 5761 MixerThread *thread1 = checkMixerThread_l(output1); 5762 MixerThread *thread2 = checkMixerThread_l(output2); 5763 5764 if (thread1 == NULL || thread2 == NULL) { 5765 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5766 return 0; 5767 } 5768 5769 audio_io_handle_t id = nextUniqueId(); 5770 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5771 thread->addOutputTrack(thread2); 5772 mPlaybackThreads.add(id, thread); 5773 // notify client processes of the new output creation 5774 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5775 return id; 5776} 5777 5778status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5779{ 5780 // keep strong reference on the playback thread so that 5781 // it is not destroyed while exit() is executed 5782 sp<PlaybackThread> thread; 5783 { 5784 Mutex::Autolock _l(mLock); 5785 thread = checkPlaybackThread_l(output); 5786 if (thread == NULL) { 5787 return BAD_VALUE; 5788 } 5789 5790 ALOGV("closeOutput() %d", output); 5791 5792 if (thread->type() == ThreadBase::MIXER) { 5793 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5794 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5795 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5796 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5797 } 5798 } 5799 } 5800 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5801 mPlaybackThreads.removeItem(output); 5802 } 5803 thread->exit(); 5804 // The thread entity (active unit of execution) is no longer running here, 5805 // but the ThreadBase container still exists. 5806 5807 if (thread->type() != ThreadBase::DUPLICATING) { 5808 AudioStreamOut *out = thread->clearOutput(); 5809 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5810 // from now on thread->mOutput is NULL 5811 out->hwDev->close_output_stream(out->hwDev, out->stream); 5812 delete out; 5813 } 5814 return NO_ERROR; 5815} 5816 5817status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5818{ 5819 Mutex::Autolock _l(mLock); 5820 PlaybackThread *thread = checkPlaybackThread_l(output); 5821 5822 if (thread == NULL) { 5823 return BAD_VALUE; 5824 } 5825 5826 ALOGV("suspendOutput() %d", output); 5827 thread->suspend(); 5828 5829 return NO_ERROR; 5830} 5831 5832status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5833{ 5834 Mutex::Autolock _l(mLock); 5835 PlaybackThread *thread = checkPlaybackThread_l(output); 5836 5837 if (thread == NULL) { 5838 return BAD_VALUE; 5839 } 5840 5841 ALOGV("restoreOutput() %d", output); 5842 5843 thread->restore(); 5844 5845 return NO_ERROR; 5846} 5847 5848audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5849 uint32_t *pSamplingRate, 5850 audio_format_t *pFormat, 5851 uint32_t *pChannels, 5852 audio_in_acoustics_t acoustics) 5853{ 5854 status_t status; 5855 RecordThread *thread = NULL; 5856 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5857 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5858 uint32_t channels = pChannels ? *pChannels : 0; 5859 uint32_t reqSamplingRate = samplingRate; 5860 audio_format_t reqFormat = format; 5861 uint32_t reqChannels = channels; 5862 audio_stream_in_t *inStream; 5863 audio_hw_device_t *inHwDev; 5864 5865 if (pDevices == NULL || *pDevices == 0) { 5866 return 0; 5867 } 5868 5869 Mutex::Autolock _l(mLock); 5870 5871 inHwDev = findSuitableHwDev_l(*pDevices); 5872 if (inHwDev == NULL) 5873 return 0; 5874 5875 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5876 &channels, &samplingRate, 5877 acoustics, 5878 &inStream); 5879 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5880 inStream, 5881 samplingRate, 5882 format, 5883 channels, 5884 acoustics, 5885 status); 5886 5887 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5888 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5889 // or stereo to mono conversions on 16 bit PCM inputs. 5890 if (inStream == NULL && status == BAD_VALUE && 5891 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5892 (samplingRate <= 2 * reqSamplingRate) && 5893 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5894 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5895 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5896 &channels, &samplingRate, 5897 acoustics, 5898 &inStream); 5899 } 5900 5901 if (inStream != NULL) { 5902 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5903 5904 audio_io_handle_t id = nextUniqueId(); 5905 // Start record thread 5906 // RecorThread require both input and output device indication to forward to audio 5907 // pre processing modules 5908 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5909 thread = new RecordThread(this, 5910 input, 5911 reqSamplingRate, 5912 reqChannels, 5913 id, 5914 device); 5915 mRecordThreads.add(id, thread); 5916 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5917 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5918 if (pFormat != NULL) *pFormat = format; 5919 if (pChannels != NULL) *pChannels = reqChannels; 5920 5921 input->stream->common.standby(&input->stream->common); 5922 5923 // notify client processes of the new input creation 5924 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5925 return id; 5926 } 5927 5928 return 0; 5929} 5930 5931status_t AudioFlinger::closeInput(audio_io_handle_t input) 5932{ 5933 // keep strong reference on the record thread so that 5934 // it is not destroyed while exit() is executed 5935 sp<RecordThread> thread; 5936 { 5937 Mutex::Autolock _l(mLock); 5938 thread = checkRecordThread_l(input); 5939 if (thread == NULL) { 5940 return BAD_VALUE; 5941 } 5942 5943 ALOGV("closeInput() %d", input); 5944 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5945 mRecordThreads.removeItem(input); 5946 } 5947 thread->exit(); 5948 // The thread entity (active unit of execution) is no longer running here, 5949 // but the ThreadBase container still exists. 5950 5951 AudioStreamIn *in = thread->clearInput(); 5952 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5953 // from now on thread->mInput is NULL 5954 in->hwDev->close_input_stream(in->hwDev, in->stream); 5955 delete in; 5956 5957 return NO_ERROR; 5958} 5959 5960status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5961{ 5962 Mutex::Autolock _l(mLock); 5963 MixerThread *dstThread = checkMixerThread_l(output); 5964 if (dstThread == NULL) { 5965 ALOGW("setStreamOutput() bad output id %d", output); 5966 return BAD_VALUE; 5967 } 5968 5969 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5970 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5971 5972 dstThread->setStreamValid(stream, true); 5973 5974 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5975 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5976 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5977 MixerThread *srcThread = (MixerThread *)thread; 5978 srcThread->setStreamValid(stream, false); 5979 srcThread->invalidateTracks(stream); 5980 } 5981 } 5982 5983 return NO_ERROR; 5984} 5985 5986 5987int AudioFlinger::newAudioSessionId() 5988{ 5989 return nextUniqueId(); 5990} 5991 5992void AudioFlinger::acquireAudioSessionId(int audioSession) 5993{ 5994 Mutex::Autolock _l(mLock); 5995 pid_t caller = IPCThreadState::self()->getCallingPid(); 5996 ALOGV("acquiring %d from %d", audioSession, caller); 5997 size_t num = mAudioSessionRefs.size(); 5998 for (size_t i = 0; i< num; i++) { 5999 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6000 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6001 ref->mCnt++; 6002 ALOGV(" incremented refcount to %d", ref->mCnt); 6003 return; 6004 } 6005 } 6006 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6007 ALOGV(" added new entry for %d", audioSession); 6008} 6009 6010void AudioFlinger::releaseAudioSessionId(int audioSession) 6011{ 6012 Mutex::Autolock _l(mLock); 6013 pid_t caller = IPCThreadState::self()->getCallingPid(); 6014 ALOGV("releasing %d from %d", audioSession, caller); 6015 size_t num = mAudioSessionRefs.size(); 6016 for (size_t i = 0; i< num; i++) { 6017 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6018 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6019 ref->mCnt--; 6020 ALOGV(" decremented refcount to %d", ref->mCnt); 6021 if (ref->mCnt == 0) { 6022 mAudioSessionRefs.removeAt(i); 6023 delete ref; 6024 purgeStaleEffects_l(); 6025 } 6026 return; 6027 } 6028 } 6029 ALOGW("session id %d not found for pid %d", audioSession, caller); 6030} 6031 6032void AudioFlinger::purgeStaleEffects_l() { 6033 6034 ALOGV("purging stale effects"); 6035 6036 Vector< sp<EffectChain> > chains; 6037 6038 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6039 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6040 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6041 sp<EffectChain> ec = t->mEffectChains[j]; 6042 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6043 chains.push(ec); 6044 } 6045 } 6046 } 6047 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6048 sp<RecordThread> t = mRecordThreads.valueAt(i); 6049 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6050 sp<EffectChain> ec = t->mEffectChains[j]; 6051 chains.push(ec); 6052 } 6053 } 6054 6055 for (size_t i = 0; i < chains.size(); i++) { 6056 sp<EffectChain> ec = chains[i]; 6057 int sessionid = ec->sessionId(); 6058 sp<ThreadBase> t = ec->mThread.promote(); 6059 if (t == 0) { 6060 continue; 6061 } 6062 size_t numsessionrefs = mAudioSessionRefs.size(); 6063 bool found = false; 6064 for (size_t k = 0; k < numsessionrefs; k++) { 6065 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6066 if (ref->mSessionid == sessionid) { 6067 ALOGV(" session %d still exists for %d with %d refs", 6068 sessionid, ref->mPid, ref->mCnt); 6069 found = true; 6070 break; 6071 } 6072 } 6073 if (!found) { 6074 // remove all effects from the chain 6075 while (ec->mEffects.size()) { 6076 sp<EffectModule> effect = ec->mEffects[0]; 6077 effect->unPin(); 6078 Mutex::Autolock _l (t->mLock); 6079 t->removeEffect_l(effect); 6080 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6081 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6082 if (handle != 0) { 6083 handle->mEffect.clear(); 6084 if (handle->mHasControl && handle->mEnabled) { 6085 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6086 } 6087 } 6088 } 6089 AudioSystem::unregisterEffect(effect->id()); 6090 } 6091 } 6092 } 6093 return; 6094} 6095 6096// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6097AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6098{ 6099 return mPlaybackThreads.valueFor(output).get(); 6100} 6101 6102// checkMixerThread_l() must be called with AudioFlinger::mLock held 6103AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6104{ 6105 PlaybackThread *thread = checkPlaybackThread_l(output); 6106 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6107} 6108 6109// checkRecordThread_l() must be called with AudioFlinger::mLock held 6110AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6111{ 6112 return mRecordThreads.valueFor(input).get(); 6113} 6114 6115uint32_t AudioFlinger::nextUniqueId() 6116{ 6117 return android_atomic_inc(&mNextUniqueId); 6118} 6119 6120AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6121{ 6122 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6123 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6124 AudioStreamOut *output = thread->getOutput(); 6125 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6126 return thread; 6127 } 6128 } 6129 return NULL; 6130} 6131 6132uint32_t AudioFlinger::primaryOutputDevice_l() const 6133{ 6134 PlaybackThread *thread = primaryPlaybackThread_l(); 6135 6136 if (thread == NULL) { 6137 return 0; 6138 } 6139 6140 return thread->device(); 6141} 6142 6143sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6144 int triggerSession, 6145 int listenerSession, 6146 sync_event_callback_t callBack, 6147 void *cookie) 6148{ 6149 Mutex::Autolock _l(mLock); 6150 6151 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6152 status_t playStatus = NAME_NOT_FOUND; 6153 status_t recStatus = NAME_NOT_FOUND; 6154 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6155 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6156 if (playStatus == NO_ERROR) { 6157 return event; 6158 } 6159 } 6160 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6161 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6162 if (recStatus == NO_ERROR) { 6163 return event; 6164 } 6165 } 6166 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6167 mPendingSyncEvents.add(event); 6168 } else { 6169 ALOGV("createSyncEvent() invalid event %d", event->type()); 6170 event.clear(); 6171 } 6172 return event; 6173} 6174 6175// ---------------------------------------------------------------------------- 6176// Effect management 6177// ---------------------------------------------------------------------------- 6178 6179 6180status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6181{ 6182 Mutex::Autolock _l(mLock); 6183 return EffectQueryNumberEffects(numEffects); 6184} 6185 6186status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6187{ 6188 Mutex::Autolock _l(mLock); 6189 return EffectQueryEffect(index, descriptor); 6190} 6191 6192status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6193 effect_descriptor_t *descriptor) const 6194{ 6195 Mutex::Autolock _l(mLock); 6196 return EffectGetDescriptor(pUuid, descriptor); 6197} 6198 6199 6200sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6201 effect_descriptor_t *pDesc, 6202 const sp<IEffectClient>& effectClient, 6203 int32_t priority, 6204 audio_io_handle_t io, 6205 int sessionId, 6206 status_t *status, 6207 int *id, 6208 int *enabled) 6209{ 6210 status_t lStatus = NO_ERROR; 6211 sp<EffectHandle> handle; 6212 effect_descriptor_t desc; 6213 6214 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6215 pid, effectClient.get(), priority, sessionId, io); 6216 6217 if (pDesc == NULL) { 6218 lStatus = BAD_VALUE; 6219 goto Exit; 6220 } 6221 6222 // check audio settings permission for global effects 6223 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6224 lStatus = PERMISSION_DENIED; 6225 goto Exit; 6226 } 6227 6228 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6229 // that can only be created by audio policy manager (running in same process) 6230 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6231 lStatus = PERMISSION_DENIED; 6232 goto Exit; 6233 } 6234 6235 if (io == 0) { 6236 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6237 // output must be specified by AudioPolicyManager when using session 6238 // AUDIO_SESSION_OUTPUT_STAGE 6239 lStatus = BAD_VALUE; 6240 goto Exit; 6241 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6242 // if the output returned by getOutputForEffect() is removed before we lock the 6243 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6244 // and we will exit safely 6245 io = AudioSystem::getOutputForEffect(&desc); 6246 } 6247 } 6248 6249 { 6250 Mutex::Autolock _l(mLock); 6251 6252 6253 if (!EffectIsNullUuid(&pDesc->uuid)) { 6254 // if uuid is specified, request effect descriptor 6255 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6256 if (lStatus < 0) { 6257 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6258 goto Exit; 6259 } 6260 } else { 6261 // if uuid is not specified, look for an available implementation 6262 // of the required type in effect factory 6263 if (EffectIsNullUuid(&pDesc->type)) { 6264 ALOGW("createEffect() no effect type"); 6265 lStatus = BAD_VALUE; 6266 goto Exit; 6267 } 6268 uint32_t numEffects = 0; 6269 effect_descriptor_t d; 6270 d.flags = 0; // prevent compiler warning 6271 bool found = false; 6272 6273 lStatus = EffectQueryNumberEffects(&numEffects); 6274 if (lStatus < 0) { 6275 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6276 goto Exit; 6277 } 6278 for (uint32_t i = 0; i < numEffects; i++) { 6279 lStatus = EffectQueryEffect(i, &desc); 6280 if (lStatus < 0) { 6281 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6282 continue; 6283 } 6284 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6285 // If matching type found save effect descriptor. If the session is 6286 // 0 and the effect is not auxiliary, continue enumeration in case 6287 // an auxiliary version of this effect type is available 6288 found = true; 6289 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6290 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6291 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6292 break; 6293 } 6294 } 6295 } 6296 if (!found) { 6297 lStatus = BAD_VALUE; 6298 ALOGW("createEffect() effect not found"); 6299 goto Exit; 6300 } 6301 // For same effect type, chose auxiliary version over insert version if 6302 // connect to output mix (Compliance to OpenSL ES) 6303 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6304 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6305 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6306 } 6307 } 6308 6309 // Do not allow auxiliary effects on a session different from 0 (output mix) 6310 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6311 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6312 lStatus = INVALID_OPERATION; 6313 goto Exit; 6314 } 6315 6316 // check recording permission for visualizer 6317 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6318 !recordingAllowed()) { 6319 lStatus = PERMISSION_DENIED; 6320 goto Exit; 6321 } 6322 6323 // return effect descriptor 6324 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6325 6326 // If output is not specified try to find a matching audio session ID in one of the 6327 // output threads. 6328 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6329 // because of code checking output when entering the function. 6330 // Note: io is never 0 when creating an effect on an input 6331 if (io == 0) { 6332 // look for the thread where the specified audio session is present 6333 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6334 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6335 io = mPlaybackThreads.keyAt(i); 6336 break; 6337 } 6338 } 6339 if (io == 0) { 6340 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6341 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6342 io = mRecordThreads.keyAt(i); 6343 break; 6344 } 6345 } 6346 } 6347 // If no output thread contains the requested session ID, default to 6348 // first output. The effect chain will be moved to the correct output 6349 // thread when a track with the same session ID is created 6350 if (io == 0 && mPlaybackThreads.size()) { 6351 io = mPlaybackThreads.keyAt(0); 6352 } 6353 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6354 } 6355 ThreadBase *thread = checkRecordThread_l(io); 6356 if (thread == NULL) { 6357 thread = checkPlaybackThread_l(io); 6358 if (thread == NULL) { 6359 ALOGE("createEffect() unknown output thread"); 6360 lStatus = BAD_VALUE; 6361 goto Exit; 6362 } 6363 } 6364 6365 sp<Client> client = registerPid_l(pid); 6366 6367 // create effect on selected output thread 6368 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6369 &desc, enabled, &lStatus); 6370 if (handle != 0 && id != NULL) { 6371 *id = handle->id(); 6372 } 6373 } 6374 6375Exit: 6376 if (status != NULL) { 6377 *status = lStatus; 6378 } 6379 return handle; 6380} 6381 6382status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6383 audio_io_handle_t dstOutput) 6384{ 6385 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6386 sessionId, srcOutput, dstOutput); 6387 Mutex::Autolock _l(mLock); 6388 if (srcOutput == dstOutput) { 6389 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6390 return NO_ERROR; 6391 } 6392 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6393 if (srcThread == NULL) { 6394 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6395 return BAD_VALUE; 6396 } 6397 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6398 if (dstThread == NULL) { 6399 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6400 return BAD_VALUE; 6401 } 6402 6403 Mutex::Autolock _dl(dstThread->mLock); 6404 Mutex::Autolock _sl(srcThread->mLock); 6405 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6406 6407 return NO_ERROR; 6408} 6409 6410// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6411status_t AudioFlinger::moveEffectChain_l(int sessionId, 6412 AudioFlinger::PlaybackThread *srcThread, 6413 AudioFlinger::PlaybackThread *dstThread, 6414 bool reRegister) 6415{ 6416 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6417 sessionId, srcThread, dstThread); 6418 6419 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6420 if (chain == 0) { 6421 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6422 sessionId, srcThread); 6423 return INVALID_OPERATION; 6424 } 6425 6426 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6427 // so that a new chain is created with correct parameters when first effect is added. This is 6428 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6429 // removed. 6430 srcThread->removeEffectChain_l(chain); 6431 6432 // transfer all effects one by one so that new effect chain is created on new thread with 6433 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6434 audio_io_handle_t dstOutput = dstThread->id(); 6435 sp<EffectChain> dstChain; 6436 uint32_t strategy = 0; // prevent compiler warning 6437 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6438 while (effect != 0) { 6439 srcThread->removeEffect_l(effect); 6440 dstThread->addEffect_l(effect); 6441 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6442 if (effect->state() == EffectModule::ACTIVE || 6443 effect->state() == EffectModule::STOPPING) { 6444 effect->start(); 6445 } 6446 // if the move request is not received from audio policy manager, the effect must be 6447 // re-registered with the new strategy and output 6448 if (dstChain == 0) { 6449 dstChain = effect->chain().promote(); 6450 if (dstChain == 0) { 6451 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6452 srcThread->addEffect_l(effect); 6453 return NO_INIT; 6454 } 6455 strategy = dstChain->strategy(); 6456 } 6457 if (reRegister) { 6458 AudioSystem::unregisterEffect(effect->id()); 6459 AudioSystem::registerEffect(&effect->desc(), 6460 dstOutput, 6461 strategy, 6462 sessionId, 6463 effect->id()); 6464 } 6465 effect = chain->getEffectFromId_l(0); 6466 } 6467 6468 return NO_ERROR; 6469} 6470 6471 6472// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6473sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6474 const sp<AudioFlinger::Client>& client, 6475 const sp<IEffectClient>& effectClient, 6476 int32_t priority, 6477 int sessionId, 6478 effect_descriptor_t *desc, 6479 int *enabled, 6480 status_t *status 6481 ) 6482{ 6483 sp<EffectModule> effect; 6484 sp<EffectHandle> handle; 6485 status_t lStatus; 6486 sp<EffectChain> chain; 6487 bool chainCreated = false; 6488 bool effectCreated = false; 6489 bool effectRegistered = false; 6490 6491 lStatus = initCheck(); 6492 if (lStatus != NO_ERROR) { 6493 ALOGW("createEffect_l() Audio driver not initialized."); 6494 goto Exit; 6495 } 6496 6497 // Do not allow effects with session ID 0 on direct output or duplicating threads 6498 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6499 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6500 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6501 desc->name, sessionId); 6502 lStatus = BAD_VALUE; 6503 goto Exit; 6504 } 6505 // Only Pre processor effects are allowed on input threads and only on input threads 6506 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6507 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6508 desc->name, desc->flags, mType); 6509 lStatus = BAD_VALUE; 6510 goto Exit; 6511 } 6512 6513 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6514 6515 { // scope for mLock 6516 Mutex::Autolock _l(mLock); 6517 6518 // check for existing effect chain with the requested audio session 6519 chain = getEffectChain_l(sessionId); 6520 if (chain == 0) { 6521 // create a new chain for this session 6522 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6523 chain = new EffectChain(this, sessionId); 6524 addEffectChain_l(chain); 6525 chain->setStrategy(getStrategyForSession_l(sessionId)); 6526 chainCreated = true; 6527 } else { 6528 effect = chain->getEffectFromDesc_l(desc); 6529 } 6530 6531 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6532 6533 if (effect == 0) { 6534 int id = mAudioFlinger->nextUniqueId(); 6535 // Check CPU and memory usage 6536 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6537 if (lStatus != NO_ERROR) { 6538 goto Exit; 6539 } 6540 effectRegistered = true; 6541 // create a new effect module if none present in the chain 6542 effect = new EffectModule(this, chain, desc, id, sessionId); 6543 lStatus = effect->status(); 6544 if (lStatus != NO_ERROR) { 6545 goto Exit; 6546 } 6547 lStatus = chain->addEffect_l(effect); 6548 if (lStatus != NO_ERROR) { 6549 goto Exit; 6550 } 6551 effectCreated = true; 6552 6553 effect->setDevice(mDevice); 6554 effect->setMode(mAudioFlinger->getMode()); 6555 } 6556 // create effect handle and connect it to effect module 6557 handle = new EffectHandle(effect, client, effectClient, priority); 6558 lStatus = effect->addHandle(handle); 6559 if (enabled != NULL) { 6560 *enabled = (int)effect->isEnabled(); 6561 } 6562 } 6563 6564Exit: 6565 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6566 Mutex::Autolock _l(mLock); 6567 if (effectCreated) { 6568 chain->removeEffect_l(effect); 6569 } 6570 if (effectRegistered) { 6571 AudioSystem::unregisterEffect(effect->id()); 6572 } 6573 if (chainCreated) { 6574 removeEffectChain_l(chain); 6575 } 6576 handle.clear(); 6577 } 6578 6579 if (status != NULL) { 6580 *status = lStatus; 6581 } 6582 return handle; 6583} 6584 6585sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6586{ 6587 sp<EffectChain> chain = getEffectChain_l(sessionId); 6588 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6589} 6590 6591// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6592// PlaybackThread::mLock held 6593status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6594{ 6595 // check for existing effect chain with the requested audio session 6596 int sessionId = effect->sessionId(); 6597 sp<EffectChain> chain = getEffectChain_l(sessionId); 6598 bool chainCreated = false; 6599 6600 if (chain == 0) { 6601 // create a new chain for this session 6602 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6603 chain = new EffectChain(this, sessionId); 6604 addEffectChain_l(chain); 6605 chain->setStrategy(getStrategyForSession_l(sessionId)); 6606 chainCreated = true; 6607 } 6608 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6609 6610 if (chain->getEffectFromId_l(effect->id()) != 0) { 6611 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6612 this, effect->desc().name, chain.get()); 6613 return BAD_VALUE; 6614 } 6615 6616 status_t status = chain->addEffect_l(effect); 6617 if (status != NO_ERROR) { 6618 if (chainCreated) { 6619 removeEffectChain_l(chain); 6620 } 6621 return status; 6622 } 6623 6624 effect->setDevice(mDevice); 6625 effect->setMode(mAudioFlinger->getMode()); 6626 return NO_ERROR; 6627} 6628 6629void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6630 6631 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6632 effect_descriptor_t desc = effect->desc(); 6633 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6634 detachAuxEffect_l(effect->id()); 6635 } 6636 6637 sp<EffectChain> chain = effect->chain().promote(); 6638 if (chain != 0) { 6639 // remove effect chain if removing last effect 6640 if (chain->removeEffect_l(effect) == 0) { 6641 removeEffectChain_l(chain); 6642 } 6643 } else { 6644 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6645 } 6646} 6647 6648void AudioFlinger::ThreadBase::lockEffectChains_l( 6649 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6650{ 6651 effectChains = mEffectChains; 6652 for (size_t i = 0; i < mEffectChains.size(); i++) { 6653 mEffectChains[i]->lock(); 6654 } 6655} 6656 6657void AudioFlinger::ThreadBase::unlockEffectChains( 6658 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6659{ 6660 for (size_t i = 0; i < effectChains.size(); i++) { 6661 effectChains[i]->unlock(); 6662 } 6663} 6664 6665sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6666{ 6667 Mutex::Autolock _l(mLock); 6668 return getEffectChain_l(sessionId); 6669} 6670 6671sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6672{ 6673 size_t size = mEffectChains.size(); 6674 for (size_t i = 0; i < size; i++) { 6675 if (mEffectChains[i]->sessionId() == sessionId) { 6676 return mEffectChains[i]; 6677 } 6678 } 6679 return 0; 6680} 6681 6682void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6683{ 6684 Mutex::Autolock _l(mLock); 6685 size_t size = mEffectChains.size(); 6686 for (size_t i = 0; i < size; i++) { 6687 mEffectChains[i]->setMode_l(mode); 6688 } 6689} 6690 6691void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6692 const wp<EffectHandle>& handle, 6693 bool unpinIfLast) { 6694 6695 Mutex::Autolock _l(mLock); 6696 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6697 // delete the effect module if removing last handle on it 6698 if (effect->removeHandle(handle) == 0) { 6699 if (!effect->isPinned() || unpinIfLast) { 6700 removeEffect_l(effect); 6701 AudioSystem::unregisterEffect(effect->id()); 6702 } 6703 } 6704} 6705 6706status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6707{ 6708 int session = chain->sessionId(); 6709 int16_t *buffer = mMixBuffer; 6710 bool ownsBuffer = false; 6711 6712 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6713 if (session > 0) { 6714 // Only one effect chain can be present in direct output thread and it uses 6715 // the mix buffer as input 6716 if (mType != DIRECT) { 6717 size_t numSamples = mFrameCount * mChannelCount; 6718 buffer = new int16_t[numSamples]; 6719 memset(buffer, 0, numSamples * sizeof(int16_t)); 6720 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6721 ownsBuffer = true; 6722 } 6723 6724 // Attach all tracks with same session ID to this chain. 6725 for (size_t i = 0; i < mTracks.size(); ++i) { 6726 sp<Track> track = mTracks[i]; 6727 if (session == track->sessionId()) { 6728 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6729 track->setMainBuffer(buffer); 6730 chain->incTrackCnt(); 6731 } 6732 } 6733 6734 // indicate all active tracks in the chain 6735 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6736 sp<Track> track = mActiveTracks[i].promote(); 6737 if (track == 0) continue; 6738 if (session == track->sessionId()) { 6739 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6740 chain->incActiveTrackCnt(); 6741 } 6742 } 6743 } 6744 6745 chain->setInBuffer(buffer, ownsBuffer); 6746 chain->setOutBuffer(mMixBuffer); 6747 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6748 // chains list in order to be processed last as it contains output stage effects 6749 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6750 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6751 // after track specific effects and before output stage 6752 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6753 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6754 // Effect chain for other sessions are inserted at beginning of effect 6755 // chains list to be processed before output mix effects. Relative order between other 6756 // sessions is not important 6757 size_t size = mEffectChains.size(); 6758 size_t i = 0; 6759 for (i = 0; i < size; i++) { 6760 if (mEffectChains[i]->sessionId() < session) break; 6761 } 6762 mEffectChains.insertAt(chain, i); 6763 checkSuspendOnAddEffectChain_l(chain); 6764 6765 return NO_ERROR; 6766} 6767 6768size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6769{ 6770 int session = chain->sessionId(); 6771 6772 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6773 6774 for (size_t i = 0; i < mEffectChains.size(); i++) { 6775 if (chain == mEffectChains[i]) { 6776 mEffectChains.removeAt(i); 6777 // detach all active tracks from the chain 6778 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6779 sp<Track> track = mActiveTracks[i].promote(); 6780 if (track == 0) continue; 6781 if (session == track->sessionId()) { 6782 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6783 chain.get(), session); 6784 chain->decActiveTrackCnt(); 6785 } 6786 } 6787 6788 // detach all tracks with same session ID from this chain 6789 for (size_t i = 0; i < mTracks.size(); ++i) { 6790 sp<Track> track = mTracks[i]; 6791 if (session == track->sessionId()) { 6792 track->setMainBuffer(mMixBuffer); 6793 chain->decTrackCnt(); 6794 } 6795 } 6796 break; 6797 } 6798 } 6799 return mEffectChains.size(); 6800} 6801 6802status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6803 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6804{ 6805 Mutex::Autolock _l(mLock); 6806 return attachAuxEffect_l(track, EffectId); 6807} 6808 6809status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6810 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6811{ 6812 status_t status = NO_ERROR; 6813 6814 if (EffectId == 0) { 6815 track->setAuxBuffer(0, NULL); 6816 } else { 6817 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6818 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6819 if (effect != 0) { 6820 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6821 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6822 } else { 6823 status = INVALID_OPERATION; 6824 } 6825 } else { 6826 status = BAD_VALUE; 6827 } 6828 } 6829 return status; 6830} 6831 6832void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6833{ 6834 for (size_t i = 0; i < mTracks.size(); ++i) { 6835 sp<Track> track = mTracks[i]; 6836 if (track->auxEffectId() == effectId) { 6837 attachAuxEffect_l(track, 0); 6838 } 6839 } 6840} 6841 6842status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6843{ 6844 // only one chain per input thread 6845 if (mEffectChains.size() != 0) { 6846 return INVALID_OPERATION; 6847 } 6848 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6849 6850 chain->setInBuffer(NULL); 6851 chain->setOutBuffer(NULL); 6852 6853 checkSuspendOnAddEffectChain_l(chain); 6854 6855 mEffectChains.add(chain); 6856 6857 return NO_ERROR; 6858} 6859 6860size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6861{ 6862 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6863 ALOGW_IF(mEffectChains.size() != 1, 6864 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6865 chain.get(), mEffectChains.size(), this); 6866 if (mEffectChains.size() == 1) { 6867 mEffectChains.removeAt(0); 6868 } 6869 return 0; 6870} 6871 6872// ---------------------------------------------------------------------------- 6873// EffectModule implementation 6874// ---------------------------------------------------------------------------- 6875 6876#undef LOG_TAG 6877#define LOG_TAG "AudioFlinger::EffectModule" 6878 6879AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6880 const wp<AudioFlinger::EffectChain>& chain, 6881 effect_descriptor_t *desc, 6882 int id, 6883 int sessionId) 6884 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6885 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6886{ 6887 ALOGV("Constructor %p", this); 6888 int lStatus; 6889 if (thread == NULL) { 6890 return; 6891 } 6892 6893 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6894 6895 // create effect engine from effect factory 6896 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6897 6898 if (mStatus != NO_ERROR) { 6899 return; 6900 } 6901 lStatus = init(); 6902 if (lStatus < 0) { 6903 mStatus = lStatus; 6904 goto Error; 6905 } 6906 6907 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6908 mPinned = true; 6909 } 6910 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6911 return; 6912Error: 6913 EffectRelease(mEffectInterface); 6914 mEffectInterface = NULL; 6915 ALOGV("Constructor Error %d", mStatus); 6916} 6917 6918AudioFlinger::EffectModule::~EffectModule() 6919{ 6920 ALOGV("Destructor %p", this); 6921 if (mEffectInterface != NULL) { 6922 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6923 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6924 sp<ThreadBase> thread = mThread.promote(); 6925 if (thread != 0) { 6926 audio_stream_t *stream = thread->stream(); 6927 if (stream != NULL) { 6928 stream->remove_audio_effect(stream, mEffectInterface); 6929 } 6930 } 6931 } 6932 // release effect engine 6933 EffectRelease(mEffectInterface); 6934 } 6935} 6936 6937status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6938{ 6939 status_t status; 6940 6941 Mutex::Autolock _l(mLock); 6942 int priority = handle->priority(); 6943 size_t size = mHandles.size(); 6944 sp<EffectHandle> h; 6945 size_t i; 6946 for (i = 0; i < size; i++) { 6947 h = mHandles[i].promote(); 6948 if (h == 0) continue; 6949 if (h->priority() <= priority) break; 6950 } 6951 // if inserted in first place, move effect control from previous owner to this handle 6952 if (i == 0) { 6953 bool enabled = false; 6954 if (h != 0) { 6955 enabled = h->enabled(); 6956 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6957 } 6958 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6959 status = NO_ERROR; 6960 } else { 6961 status = ALREADY_EXISTS; 6962 } 6963 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6964 mHandles.insertAt(handle, i); 6965 return status; 6966} 6967 6968size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6969{ 6970 Mutex::Autolock _l(mLock); 6971 size_t size = mHandles.size(); 6972 size_t i; 6973 for (i = 0; i < size; i++) { 6974 if (mHandles[i] == handle) break; 6975 } 6976 if (i == size) { 6977 return size; 6978 } 6979 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6980 6981 bool enabled = false; 6982 EffectHandle *hdl = handle.unsafe_get(); 6983 if (hdl != NULL) { 6984 ALOGV("removeHandle() unsafe_get OK"); 6985 enabled = hdl->enabled(); 6986 } 6987 mHandles.removeAt(i); 6988 size = mHandles.size(); 6989 // if removed from first place, move effect control from this handle to next in line 6990 if (i == 0 && size != 0) { 6991 sp<EffectHandle> h = mHandles[0].promote(); 6992 if (h != 0) { 6993 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6994 } 6995 } 6996 6997 // Prevent calls to process() and other functions on effect interface from now on. 6998 // The effect engine will be released by the destructor when the last strong reference on 6999 // this object is released which can happen after next process is called. 7000 if (size == 0 && !mPinned) { 7001 mState = DESTROYED; 7002 } 7003 7004 return size; 7005} 7006 7007sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7008{ 7009 Mutex::Autolock _l(mLock); 7010 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7011} 7012 7013void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7014{ 7015 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7016 // keep a strong reference on this EffectModule to avoid calling the 7017 // destructor before we exit 7018 sp<EffectModule> keep(this); 7019 { 7020 sp<ThreadBase> thread = mThread.promote(); 7021 if (thread != 0) { 7022 thread->disconnectEffect(keep, handle, unpinIfLast); 7023 } 7024 } 7025} 7026 7027void AudioFlinger::EffectModule::updateState() { 7028 Mutex::Autolock _l(mLock); 7029 7030 switch (mState) { 7031 case RESTART: 7032 reset_l(); 7033 // FALL THROUGH 7034 7035 case STARTING: 7036 // clear auxiliary effect input buffer for next accumulation 7037 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7038 memset(mConfig.inputCfg.buffer.raw, 7039 0, 7040 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7041 } 7042 start_l(); 7043 mState = ACTIVE; 7044 break; 7045 case STOPPING: 7046 stop_l(); 7047 mDisableWaitCnt = mMaxDisableWaitCnt; 7048 mState = STOPPED; 7049 break; 7050 case STOPPED: 7051 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7052 // turn off sequence. 7053 if (--mDisableWaitCnt == 0) { 7054 reset_l(); 7055 mState = IDLE; 7056 } 7057 break; 7058 default: //IDLE , ACTIVE, DESTROYED 7059 break; 7060 } 7061} 7062 7063void AudioFlinger::EffectModule::process() 7064{ 7065 Mutex::Autolock _l(mLock); 7066 7067 if (mState == DESTROYED || mEffectInterface == NULL || 7068 mConfig.inputCfg.buffer.raw == NULL || 7069 mConfig.outputCfg.buffer.raw == NULL) { 7070 return; 7071 } 7072 7073 if (isProcessEnabled()) { 7074 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7075 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7076 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7077 mConfig.inputCfg.buffer.s32, 7078 mConfig.inputCfg.buffer.frameCount/2); 7079 } 7080 7081 // do the actual processing in the effect engine 7082 int ret = (*mEffectInterface)->process(mEffectInterface, 7083 &mConfig.inputCfg.buffer, 7084 &mConfig.outputCfg.buffer); 7085 7086 // force transition to IDLE state when engine is ready 7087 if (mState == STOPPED && ret == -ENODATA) { 7088 mDisableWaitCnt = 1; 7089 } 7090 7091 // clear auxiliary effect input buffer for next accumulation 7092 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7093 memset(mConfig.inputCfg.buffer.raw, 0, 7094 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7095 } 7096 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7097 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7098 // If an insert effect is idle and input buffer is different from output buffer, 7099 // accumulate input onto output 7100 sp<EffectChain> chain = mChain.promote(); 7101 if (chain != 0 && chain->activeTrackCnt() != 0) { 7102 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7103 int16_t *in = mConfig.inputCfg.buffer.s16; 7104 int16_t *out = mConfig.outputCfg.buffer.s16; 7105 for (size_t i = 0; i < frameCnt; i++) { 7106 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7107 } 7108 } 7109 } 7110} 7111 7112void AudioFlinger::EffectModule::reset_l() 7113{ 7114 if (mEffectInterface == NULL) { 7115 return; 7116 } 7117 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7118} 7119 7120status_t AudioFlinger::EffectModule::configure() 7121{ 7122 uint32_t channels; 7123 if (mEffectInterface == NULL) { 7124 return NO_INIT; 7125 } 7126 7127 sp<ThreadBase> thread = mThread.promote(); 7128 if (thread == 0) { 7129 return DEAD_OBJECT; 7130 } 7131 7132 // TODO: handle configuration of effects replacing track process 7133 if (thread->channelCount() == 1) { 7134 channels = AUDIO_CHANNEL_OUT_MONO; 7135 } else { 7136 channels = AUDIO_CHANNEL_OUT_STEREO; 7137 } 7138 7139 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7140 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7141 } else { 7142 mConfig.inputCfg.channels = channels; 7143 } 7144 mConfig.outputCfg.channels = channels; 7145 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7146 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7147 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7148 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7149 mConfig.inputCfg.bufferProvider.cookie = NULL; 7150 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7151 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7152 mConfig.outputCfg.bufferProvider.cookie = NULL; 7153 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7154 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7155 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7156 // Insert effect: 7157 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7158 // always overwrites output buffer: input buffer == output buffer 7159 // - in other sessions: 7160 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7161 // other effect: overwrites output buffer: input buffer == output buffer 7162 // Auxiliary effect: 7163 // accumulates in output buffer: input buffer != output buffer 7164 // Therefore: accumulate <=> input buffer != output buffer 7165 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7166 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7167 } else { 7168 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7169 } 7170 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7171 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7172 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7173 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7174 7175 ALOGV("configure() %p thread %p buffer %p framecount %d", 7176 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7177 7178 status_t cmdStatus; 7179 uint32_t size = sizeof(int); 7180 status_t status = (*mEffectInterface)->command(mEffectInterface, 7181 EFFECT_CMD_SET_CONFIG, 7182 sizeof(effect_config_t), 7183 &mConfig, 7184 &size, 7185 &cmdStatus); 7186 if (status == 0) { 7187 status = cmdStatus; 7188 } 7189 7190 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7191 (1000 * mConfig.outputCfg.buffer.frameCount); 7192 7193 return status; 7194} 7195 7196status_t AudioFlinger::EffectModule::init() 7197{ 7198 Mutex::Autolock _l(mLock); 7199 if (mEffectInterface == NULL) { 7200 return NO_INIT; 7201 } 7202 status_t cmdStatus; 7203 uint32_t size = sizeof(status_t); 7204 status_t status = (*mEffectInterface)->command(mEffectInterface, 7205 EFFECT_CMD_INIT, 7206 0, 7207 NULL, 7208 &size, 7209 &cmdStatus); 7210 if (status == 0) { 7211 status = cmdStatus; 7212 } 7213 return status; 7214} 7215 7216status_t AudioFlinger::EffectModule::start() 7217{ 7218 Mutex::Autolock _l(mLock); 7219 return start_l(); 7220} 7221 7222status_t AudioFlinger::EffectModule::start_l() 7223{ 7224 if (mEffectInterface == NULL) { 7225 return NO_INIT; 7226 } 7227 status_t cmdStatus; 7228 uint32_t size = sizeof(status_t); 7229 status_t status = (*mEffectInterface)->command(mEffectInterface, 7230 EFFECT_CMD_ENABLE, 7231 0, 7232 NULL, 7233 &size, 7234 &cmdStatus); 7235 if (status == 0) { 7236 status = cmdStatus; 7237 } 7238 if (status == 0 && 7239 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7240 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7241 sp<ThreadBase> thread = mThread.promote(); 7242 if (thread != 0) { 7243 audio_stream_t *stream = thread->stream(); 7244 if (stream != NULL) { 7245 stream->add_audio_effect(stream, mEffectInterface); 7246 } 7247 } 7248 } 7249 return status; 7250} 7251 7252status_t AudioFlinger::EffectModule::stop() 7253{ 7254 Mutex::Autolock _l(mLock); 7255 return stop_l(); 7256} 7257 7258status_t AudioFlinger::EffectModule::stop_l() 7259{ 7260 if (mEffectInterface == NULL) { 7261 return NO_INIT; 7262 } 7263 status_t cmdStatus; 7264 uint32_t size = sizeof(status_t); 7265 status_t status = (*mEffectInterface)->command(mEffectInterface, 7266 EFFECT_CMD_DISABLE, 7267 0, 7268 NULL, 7269 &size, 7270 &cmdStatus); 7271 if (status == 0) { 7272 status = cmdStatus; 7273 } 7274 if (status == 0 && 7275 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7276 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7277 sp<ThreadBase> thread = mThread.promote(); 7278 if (thread != 0) { 7279 audio_stream_t *stream = thread->stream(); 7280 if (stream != NULL) { 7281 stream->remove_audio_effect(stream, mEffectInterface); 7282 } 7283 } 7284 } 7285 return status; 7286} 7287 7288status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7289 uint32_t cmdSize, 7290 void *pCmdData, 7291 uint32_t *replySize, 7292 void *pReplyData) 7293{ 7294 Mutex::Autolock _l(mLock); 7295// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7296 7297 if (mState == DESTROYED || mEffectInterface == NULL) { 7298 return NO_INIT; 7299 } 7300 status_t status = (*mEffectInterface)->command(mEffectInterface, 7301 cmdCode, 7302 cmdSize, 7303 pCmdData, 7304 replySize, 7305 pReplyData); 7306 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7307 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7308 for (size_t i = 1; i < mHandles.size(); i++) { 7309 sp<EffectHandle> h = mHandles[i].promote(); 7310 if (h != 0) { 7311 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7312 } 7313 } 7314 } 7315 return status; 7316} 7317 7318status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7319{ 7320 7321 Mutex::Autolock _l(mLock); 7322 ALOGV("setEnabled %p enabled %d", this, enabled); 7323 7324 if (enabled != isEnabled()) { 7325 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7326 if (enabled && status != NO_ERROR) { 7327 return status; 7328 } 7329 7330 switch (mState) { 7331 // going from disabled to enabled 7332 case IDLE: 7333 mState = STARTING; 7334 break; 7335 case STOPPED: 7336 mState = RESTART; 7337 break; 7338 case STOPPING: 7339 mState = ACTIVE; 7340 break; 7341 7342 // going from enabled to disabled 7343 case RESTART: 7344 mState = STOPPED; 7345 break; 7346 case STARTING: 7347 mState = IDLE; 7348 break; 7349 case ACTIVE: 7350 mState = STOPPING; 7351 break; 7352 case DESTROYED: 7353 return NO_ERROR; // simply ignore as we are being destroyed 7354 } 7355 for (size_t i = 1; i < mHandles.size(); i++) { 7356 sp<EffectHandle> h = mHandles[i].promote(); 7357 if (h != 0) { 7358 h->setEnabled(enabled); 7359 } 7360 } 7361 } 7362 return NO_ERROR; 7363} 7364 7365bool AudioFlinger::EffectModule::isEnabled() const 7366{ 7367 switch (mState) { 7368 case RESTART: 7369 case STARTING: 7370 case ACTIVE: 7371 return true; 7372 case IDLE: 7373 case STOPPING: 7374 case STOPPED: 7375 case DESTROYED: 7376 default: 7377 return false; 7378 } 7379} 7380 7381bool AudioFlinger::EffectModule::isProcessEnabled() const 7382{ 7383 switch (mState) { 7384 case RESTART: 7385 case ACTIVE: 7386 case STOPPING: 7387 case STOPPED: 7388 return true; 7389 case IDLE: 7390 case STARTING: 7391 case DESTROYED: 7392 default: 7393 return false; 7394 } 7395} 7396 7397status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7398{ 7399 Mutex::Autolock _l(mLock); 7400 status_t status = NO_ERROR; 7401 7402 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7403 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7404 if (isProcessEnabled() && 7405 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7406 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7407 status_t cmdStatus; 7408 uint32_t volume[2]; 7409 uint32_t *pVolume = NULL; 7410 uint32_t size = sizeof(volume); 7411 volume[0] = *left; 7412 volume[1] = *right; 7413 if (controller) { 7414 pVolume = volume; 7415 } 7416 status = (*mEffectInterface)->command(mEffectInterface, 7417 EFFECT_CMD_SET_VOLUME, 7418 size, 7419 volume, 7420 &size, 7421 pVolume); 7422 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7423 *left = volume[0]; 7424 *right = volume[1]; 7425 } 7426 } 7427 return status; 7428} 7429 7430status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7431{ 7432 Mutex::Autolock _l(mLock); 7433 status_t status = NO_ERROR; 7434 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7435 // audio pre processing modules on RecordThread can receive both output and 7436 // input device indication in the same call 7437 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7438 if (dev) { 7439 status_t cmdStatus; 7440 uint32_t size = sizeof(status_t); 7441 7442 status = (*mEffectInterface)->command(mEffectInterface, 7443 EFFECT_CMD_SET_DEVICE, 7444 sizeof(uint32_t), 7445 &dev, 7446 &size, 7447 &cmdStatus); 7448 if (status == NO_ERROR) { 7449 status = cmdStatus; 7450 } 7451 } 7452 dev = device & AUDIO_DEVICE_IN_ALL; 7453 if (dev) { 7454 status_t cmdStatus; 7455 uint32_t size = sizeof(status_t); 7456 7457 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7458 EFFECT_CMD_SET_INPUT_DEVICE, 7459 sizeof(uint32_t), 7460 &dev, 7461 &size, 7462 &cmdStatus); 7463 if (status2 == NO_ERROR) { 7464 status2 = cmdStatus; 7465 } 7466 if (status == NO_ERROR) { 7467 status = status2; 7468 } 7469 } 7470 } 7471 return status; 7472} 7473 7474status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7475{ 7476 Mutex::Autolock _l(mLock); 7477 status_t status = NO_ERROR; 7478 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7479 status_t cmdStatus; 7480 uint32_t size = sizeof(status_t); 7481 status = (*mEffectInterface)->command(mEffectInterface, 7482 EFFECT_CMD_SET_AUDIO_MODE, 7483 sizeof(audio_mode_t), 7484 &mode, 7485 &size, 7486 &cmdStatus); 7487 if (status == NO_ERROR) { 7488 status = cmdStatus; 7489 } 7490 } 7491 return status; 7492} 7493 7494void AudioFlinger::EffectModule::setSuspended(bool suspended) 7495{ 7496 Mutex::Autolock _l(mLock); 7497 mSuspended = suspended; 7498} 7499 7500bool AudioFlinger::EffectModule::suspended() const 7501{ 7502 Mutex::Autolock _l(mLock); 7503 return mSuspended; 7504} 7505 7506status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7507{ 7508 const size_t SIZE = 256; 7509 char buffer[SIZE]; 7510 String8 result; 7511 7512 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7513 result.append(buffer); 7514 7515 bool locked = tryLock(mLock); 7516 // failed to lock - AudioFlinger is probably deadlocked 7517 if (!locked) { 7518 result.append("\t\tCould not lock Fx mutex:\n"); 7519 } 7520 7521 result.append("\t\tSession Status State Engine:\n"); 7522 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7523 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7524 result.append(buffer); 7525 7526 result.append("\t\tDescriptor:\n"); 7527 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7528 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7529 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7530 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7531 result.append(buffer); 7532 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7533 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7534 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7535 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7536 result.append(buffer); 7537 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7538 mDescriptor.apiVersion, 7539 mDescriptor.flags); 7540 result.append(buffer); 7541 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7542 mDescriptor.name); 7543 result.append(buffer); 7544 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7545 mDescriptor.implementor); 7546 result.append(buffer); 7547 7548 result.append("\t\t- Input configuration:\n"); 7549 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7550 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7551 (uint32_t)mConfig.inputCfg.buffer.raw, 7552 mConfig.inputCfg.buffer.frameCount, 7553 mConfig.inputCfg.samplingRate, 7554 mConfig.inputCfg.channels, 7555 mConfig.inputCfg.format); 7556 result.append(buffer); 7557 7558 result.append("\t\t- Output configuration:\n"); 7559 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7560 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7561 (uint32_t)mConfig.outputCfg.buffer.raw, 7562 mConfig.outputCfg.buffer.frameCount, 7563 mConfig.outputCfg.samplingRate, 7564 mConfig.outputCfg.channels, 7565 mConfig.outputCfg.format); 7566 result.append(buffer); 7567 7568 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7569 result.append(buffer); 7570 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7571 for (size_t i = 0; i < mHandles.size(); ++i) { 7572 sp<EffectHandle> handle = mHandles[i].promote(); 7573 if (handle != 0) { 7574 handle->dump(buffer, SIZE); 7575 result.append(buffer); 7576 } 7577 } 7578 7579 result.append("\n"); 7580 7581 write(fd, result.string(), result.length()); 7582 7583 if (locked) { 7584 mLock.unlock(); 7585 } 7586 7587 return NO_ERROR; 7588} 7589 7590// ---------------------------------------------------------------------------- 7591// EffectHandle implementation 7592// ---------------------------------------------------------------------------- 7593 7594#undef LOG_TAG 7595#define LOG_TAG "AudioFlinger::EffectHandle" 7596 7597AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7598 const sp<AudioFlinger::Client>& client, 7599 const sp<IEffectClient>& effectClient, 7600 int32_t priority) 7601 : BnEffect(), 7602 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7603 mPriority(priority), mHasControl(false), mEnabled(false) 7604{ 7605 ALOGV("constructor %p", this); 7606 7607 if (client == 0) { 7608 return; 7609 } 7610 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7611 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7612 if (mCblkMemory != 0) { 7613 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7614 7615 if (mCblk != NULL) { 7616 new(mCblk) effect_param_cblk_t(); 7617 mBuffer = (uint8_t *)mCblk + bufOffset; 7618 } 7619 } else { 7620 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7621 return; 7622 } 7623} 7624 7625AudioFlinger::EffectHandle::~EffectHandle() 7626{ 7627 ALOGV("Destructor %p", this); 7628 disconnect(false); 7629 ALOGV("Destructor DONE %p", this); 7630} 7631 7632status_t AudioFlinger::EffectHandle::enable() 7633{ 7634 ALOGV("enable %p", this); 7635 if (!mHasControl) return INVALID_OPERATION; 7636 if (mEffect == 0) return DEAD_OBJECT; 7637 7638 if (mEnabled) { 7639 return NO_ERROR; 7640 } 7641 7642 mEnabled = true; 7643 7644 sp<ThreadBase> thread = mEffect->thread().promote(); 7645 if (thread != 0) { 7646 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7647 } 7648 7649 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7650 if (mEffect->suspended()) { 7651 return NO_ERROR; 7652 } 7653 7654 status_t status = mEffect->setEnabled(true); 7655 if (status != NO_ERROR) { 7656 if (thread != 0) { 7657 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7658 } 7659 mEnabled = false; 7660 } 7661 return status; 7662} 7663 7664status_t AudioFlinger::EffectHandle::disable() 7665{ 7666 ALOGV("disable %p", this); 7667 if (!mHasControl) return INVALID_OPERATION; 7668 if (mEffect == 0) return DEAD_OBJECT; 7669 7670 if (!mEnabled) { 7671 return NO_ERROR; 7672 } 7673 mEnabled = false; 7674 7675 if (mEffect->suspended()) { 7676 return NO_ERROR; 7677 } 7678 7679 status_t status = mEffect->setEnabled(false); 7680 7681 sp<ThreadBase> thread = mEffect->thread().promote(); 7682 if (thread != 0) { 7683 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7684 } 7685 7686 return status; 7687} 7688 7689void AudioFlinger::EffectHandle::disconnect() 7690{ 7691 disconnect(true); 7692} 7693 7694void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7695{ 7696 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7697 if (mEffect == 0) { 7698 return; 7699 } 7700 mEffect->disconnect(this, unpinIfLast); 7701 7702 if (mHasControl && mEnabled) { 7703 sp<ThreadBase> thread = mEffect->thread().promote(); 7704 if (thread != 0) { 7705 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7706 } 7707 } 7708 7709 // release sp on module => module destructor can be called now 7710 mEffect.clear(); 7711 if (mClient != 0) { 7712 if (mCblk != NULL) { 7713 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7714 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7715 } 7716 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7717 // Client destructor must run with AudioFlinger mutex locked 7718 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7719 mClient.clear(); 7720 } 7721} 7722 7723status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7724 uint32_t cmdSize, 7725 void *pCmdData, 7726 uint32_t *replySize, 7727 void *pReplyData) 7728{ 7729// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7730// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7731 7732 // only get parameter command is permitted for applications not controlling the effect 7733 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7734 return INVALID_OPERATION; 7735 } 7736 if (mEffect == 0) return DEAD_OBJECT; 7737 if (mClient == 0) return INVALID_OPERATION; 7738 7739 // handle commands that are not forwarded transparently to effect engine 7740 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7741 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7742 // no risk to block the whole media server process or mixer threads is we are stuck here 7743 Mutex::Autolock _l(mCblk->lock); 7744 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7745 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7746 mCblk->serverIndex = 0; 7747 mCblk->clientIndex = 0; 7748 return BAD_VALUE; 7749 } 7750 status_t status = NO_ERROR; 7751 while (mCblk->serverIndex < mCblk->clientIndex) { 7752 int reply; 7753 uint32_t rsize = sizeof(int); 7754 int *p = (int *)(mBuffer + mCblk->serverIndex); 7755 int size = *p++; 7756 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7757 ALOGW("command(): invalid parameter block size"); 7758 break; 7759 } 7760 effect_param_t *param = (effect_param_t *)p; 7761 if (param->psize == 0 || param->vsize == 0) { 7762 ALOGW("command(): null parameter or value size"); 7763 mCblk->serverIndex += size; 7764 continue; 7765 } 7766 uint32_t psize = sizeof(effect_param_t) + 7767 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7768 param->vsize; 7769 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7770 psize, 7771 p, 7772 &rsize, 7773 &reply); 7774 // stop at first error encountered 7775 if (ret != NO_ERROR) { 7776 status = ret; 7777 *(int *)pReplyData = reply; 7778 break; 7779 } else if (reply != NO_ERROR) { 7780 *(int *)pReplyData = reply; 7781 break; 7782 } 7783 mCblk->serverIndex += size; 7784 } 7785 mCblk->serverIndex = 0; 7786 mCblk->clientIndex = 0; 7787 return status; 7788 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7789 *(int *)pReplyData = NO_ERROR; 7790 return enable(); 7791 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7792 *(int *)pReplyData = NO_ERROR; 7793 return disable(); 7794 } 7795 7796 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7797} 7798 7799void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7800{ 7801 ALOGV("setControl %p control %d", this, hasControl); 7802 7803 mHasControl = hasControl; 7804 mEnabled = enabled; 7805 7806 if (signal && mEffectClient != 0) { 7807 mEffectClient->controlStatusChanged(hasControl); 7808 } 7809} 7810 7811void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7812 uint32_t cmdSize, 7813 void *pCmdData, 7814 uint32_t replySize, 7815 void *pReplyData) 7816{ 7817 if (mEffectClient != 0) { 7818 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7819 } 7820} 7821 7822 7823 7824void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7825{ 7826 if (mEffectClient != 0) { 7827 mEffectClient->enableStatusChanged(enabled); 7828 } 7829} 7830 7831status_t AudioFlinger::EffectHandle::onTransact( 7832 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7833{ 7834 return BnEffect::onTransact(code, data, reply, flags); 7835} 7836 7837 7838void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7839{ 7840 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7841 7842 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7843 (mClient == 0) ? getpid_cached : mClient->pid(), 7844 mPriority, 7845 mHasControl, 7846 !locked, 7847 mCblk ? mCblk->clientIndex : 0, 7848 mCblk ? mCblk->serverIndex : 0 7849 ); 7850 7851 if (locked) { 7852 mCblk->lock.unlock(); 7853 } 7854} 7855 7856#undef LOG_TAG 7857#define LOG_TAG "AudioFlinger::EffectChain" 7858 7859AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7860 int sessionId) 7861 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7862 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7863 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7864{ 7865 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7866 if (thread == NULL) { 7867 return; 7868 } 7869 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7870 thread->frameCount(); 7871} 7872 7873AudioFlinger::EffectChain::~EffectChain() 7874{ 7875 if (mOwnInBuffer) { 7876 delete mInBuffer; 7877 } 7878 7879} 7880 7881// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7882sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7883{ 7884 size_t size = mEffects.size(); 7885 7886 for (size_t i = 0; i < size; i++) { 7887 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7888 return mEffects[i]; 7889 } 7890 } 7891 return 0; 7892} 7893 7894// getEffectFromId_l() must be called with ThreadBase::mLock held 7895sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7896{ 7897 size_t size = mEffects.size(); 7898 7899 for (size_t i = 0; i < size; i++) { 7900 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7901 if (id == 0 || mEffects[i]->id() == id) { 7902 return mEffects[i]; 7903 } 7904 } 7905 return 0; 7906} 7907 7908// getEffectFromType_l() must be called with ThreadBase::mLock held 7909sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7910 const effect_uuid_t *type) 7911{ 7912 size_t size = mEffects.size(); 7913 7914 for (size_t i = 0; i < size; i++) { 7915 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7916 return mEffects[i]; 7917 } 7918 } 7919 return 0; 7920} 7921 7922// Must be called with EffectChain::mLock locked 7923void AudioFlinger::EffectChain::process_l() 7924{ 7925 sp<ThreadBase> thread = mThread.promote(); 7926 if (thread == 0) { 7927 ALOGW("process_l(): cannot promote mixer thread"); 7928 return; 7929 } 7930 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7931 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7932 // always process effects unless no more tracks are on the session and the effect tail 7933 // has been rendered 7934 bool doProcess = true; 7935 if (!isGlobalSession) { 7936 bool tracksOnSession = (trackCnt() != 0); 7937 7938 if (!tracksOnSession && mTailBufferCount == 0) { 7939 doProcess = false; 7940 } 7941 7942 if (activeTrackCnt() == 0) { 7943 // if no track is active and the effect tail has not been rendered, 7944 // the input buffer must be cleared here as the mixer process will not do it 7945 if (tracksOnSession || mTailBufferCount > 0) { 7946 size_t numSamples = thread->frameCount() * thread->channelCount(); 7947 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7948 if (mTailBufferCount > 0) { 7949 mTailBufferCount--; 7950 } 7951 } 7952 } 7953 } 7954 7955 size_t size = mEffects.size(); 7956 if (doProcess) { 7957 for (size_t i = 0; i < size; i++) { 7958 mEffects[i]->process(); 7959 } 7960 } 7961 for (size_t i = 0; i < size; i++) { 7962 mEffects[i]->updateState(); 7963 } 7964} 7965 7966// addEffect_l() must be called with PlaybackThread::mLock held 7967status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7968{ 7969 effect_descriptor_t desc = effect->desc(); 7970 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7971 7972 Mutex::Autolock _l(mLock); 7973 effect->setChain(this); 7974 sp<ThreadBase> thread = mThread.promote(); 7975 if (thread == 0) { 7976 return NO_INIT; 7977 } 7978 effect->setThread(thread); 7979 7980 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7981 // Auxiliary effects are inserted at the beginning of mEffects vector as 7982 // they are processed first and accumulated in chain input buffer 7983 mEffects.insertAt(effect, 0); 7984 7985 // the input buffer for auxiliary effect contains mono samples in 7986 // 32 bit format. This is to avoid saturation in AudoMixer 7987 // accumulation stage. Saturation is done in EffectModule::process() before 7988 // calling the process in effect engine 7989 size_t numSamples = thread->frameCount(); 7990 int32_t *buffer = new int32_t[numSamples]; 7991 memset(buffer, 0, numSamples * sizeof(int32_t)); 7992 effect->setInBuffer((int16_t *)buffer); 7993 // auxiliary effects output samples to chain input buffer for further processing 7994 // by insert effects 7995 effect->setOutBuffer(mInBuffer); 7996 } else { 7997 // Insert effects are inserted at the end of mEffects vector as they are processed 7998 // after track and auxiliary effects. 7999 // Insert effect order as a function of indicated preference: 8000 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8001 // another effect is present 8002 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8003 // last effect claiming first position 8004 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8005 // first effect claiming last position 8006 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8007 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8008 // already present 8009 8010 size_t size = mEffects.size(); 8011 size_t idx_insert = size; 8012 ssize_t idx_insert_first = -1; 8013 ssize_t idx_insert_last = -1; 8014 8015 for (size_t i = 0; i < size; i++) { 8016 effect_descriptor_t d = mEffects[i]->desc(); 8017 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8018 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8019 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8020 // check invalid effect chaining combinations 8021 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8022 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8023 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8024 return INVALID_OPERATION; 8025 } 8026 // remember position of first insert effect and by default 8027 // select this as insert position for new effect 8028 if (idx_insert == size) { 8029 idx_insert = i; 8030 } 8031 // remember position of last insert effect claiming 8032 // first position 8033 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8034 idx_insert_first = i; 8035 } 8036 // remember position of first insert effect claiming 8037 // last position 8038 if (iPref == EFFECT_FLAG_INSERT_LAST && 8039 idx_insert_last == -1) { 8040 idx_insert_last = i; 8041 } 8042 } 8043 } 8044 8045 // modify idx_insert from first position if needed 8046 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8047 if (idx_insert_last != -1) { 8048 idx_insert = idx_insert_last; 8049 } else { 8050 idx_insert = size; 8051 } 8052 } else { 8053 if (idx_insert_first != -1) { 8054 idx_insert = idx_insert_first + 1; 8055 } 8056 } 8057 8058 // always read samples from chain input buffer 8059 effect->setInBuffer(mInBuffer); 8060 8061 // if last effect in the chain, output samples to chain 8062 // output buffer, otherwise to chain input buffer 8063 if (idx_insert == size) { 8064 if (idx_insert != 0) { 8065 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8066 mEffects[idx_insert-1]->configure(); 8067 } 8068 effect->setOutBuffer(mOutBuffer); 8069 } else { 8070 effect->setOutBuffer(mInBuffer); 8071 } 8072 mEffects.insertAt(effect, idx_insert); 8073 8074 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8075 } 8076 effect->configure(); 8077 return NO_ERROR; 8078} 8079 8080// removeEffect_l() must be called with PlaybackThread::mLock held 8081size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8082{ 8083 Mutex::Autolock _l(mLock); 8084 size_t size = mEffects.size(); 8085 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8086 8087 for (size_t i = 0; i < size; i++) { 8088 if (effect == mEffects[i]) { 8089 // calling stop here will remove pre-processing effect from the audio HAL. 8090 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8091 // the middle of a read from audio HAL 8092 if (mEffects[i]->state() == EffectModule::ACTIVE || 8093 mEffects[i]->state() == EffectModule::STOPPING) { 8094 mEffects[i]->stop(); 8095 } 8096 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8097 delete[] effect->inBuffer(); 8098 } else { 8099 if (i == size - 1 && i != 0) { 8100 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8101 mEffects[i - 1]->configure(); 8102 } 8103 } 8104 mEffects.removeAt(i); 8105 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8106 break; 8107 } 8108 } 8109 8110 return mEffects.size(); 8111} 8112 8113// setDevice_l() must be called with PlaybackThread::mLock held 8114void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8115{ 8116 size_t size = mEffects.size(); 8117 for (size_t i = 0; i < size; i++) { 8118 mEffects[i]->setDevice(device); 8119 } 8120} 8121 8122// setMode_l() must be called with PlaybackThread::mLock held 8123void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8124{ 8125 size_t size = mEffects.size(); 8126 for (size_t i = 0; i < size; i++) { 8127 mEffects[i]->setMode(mode); 8128 } 8129} 8130 8131// setVolume_l() must be called with PlaybackThread::mLock held 8132bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8133{ 8134 uint32_t newLeft = *left; 8135 uint32_t newRight = *right; 8136 bool hasControl = false; 8137 int ctrlIdx = -1; 8138 size_t size = mEffects.size(); 8139 8140 // first update volume controller 8141 for (size_t i = size; i > 0; i--) { 8142 if (mEffects[i - 1]->isProcessEnabled() && 8143 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8144 ctrlIdx = i - 1; 8145 hasControl = true; 8146 break; 8147 } 8148 } 8149 8150 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8151 if (hasControl) { 8152 *left = mNewLeftVolume; 8153 *right = mNewRightVolume; 8154 } 8155 return hasControl; 8156 } 8157 8158 mVolumeCtrlIdx = ctrlIdx; 8159 mLeftVolume = newLeft; 8160 mRightVolume = newRight; 8161 8162 // second get volume update from volume controller 8163 if (ctrlIdx >= 0) { 8164 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8165 mNewLeftVolume = newLeft; 8166 mNewRightVolume = newRight; 8167 } 8168 // then indicate volume to all other effects in chain. 8169 // Pass altered volume to effects before volume controller 8170 // and requested volume to effects after controller 8171 uint32_t lVol = newLeft; 8172 uint32_t rVol = newRight; 8173 8174 for (size_t i = 0; i < size; i++) { 8175 if ((int)i == ctrlIdx) continue; 8176 // this also works for ctrlIdx == -1 when there is no volume controller 8177 if ((int)i > ctrlIdx) { 8178 lVol = *left; 8179 rVol = *right; 8180 } 8181 mEffects[i]->setVolume(&lVol, &rVol, false); 8182 } 8183 *left = newLeft; 8184 *right = newRight; 8185 8186 return hasControl; 8187} 8188 8189status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8190{ 8191 const size_t SIZE = 256; 8192 char buffer[SIZE]; 8193 String8 result; 8194 8195 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8196 result.append(buffer); 8197 8198 bool locked = tryLock(mLock); 8199 // failed to lock - AudioFlinger is probably deadlocked 8200 if (!locked) { 8201 result.append("\tCould not lock mutex:\n"); 8202 } 8203 8204 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8205 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8206 mEffects.size(), 8207 (uint32_t)mInBuffer, 8208 (uint32_t)mOutBuffer, 8209 mActiveTrackCnt); 8210 result.append(buffer); 8211 write(fd, result.string(), result.size()); 8212 8213 for (size_t i = 0; i < mEffects.size(); ++i) { 8214 sp<EffectModule> effect = mEffects[i]; 8215 if (effect != 0) { 8216 effect->dump(fd, args); 8217 } 8218 } 8219 8220 if (locked) { 8221 mLock.unlock(); 8222 } 8223 8224 return NO_ERROR; 8225} 8226 8227// must be called with ThreadBase::mLock held 8228void AudioFlinger::EffectChain::setEffectSuspended_l( 8229 const effect_uuid_t *type, bool suspend) 8230{ 8231 sp<SuspendedEffectDesc> desc; 8232 // use effect type UUID timelow as key as there is no real risk of identical 8233 // timeLow fields among effect type UUIDs. 8234 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8235 if (suspend) { 8236 if (index >= 0) { 8237 desc = mSuspendedEffects.valueAt(index); 8238 } else { 8239 desc = new SuspendedEffectDesc(); 8240 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8241 mSuspendedEffects.add(type->timeLow, desc); 8242 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8243 } 8244 if (desc->mRefCount++ == 0) { 8245 sp<EffectModule> effect = getEffectIfEnabled(type); 8246 if (effect != 0) { 8247 desc->mEffect = effect; 8248 effect->setSuspended(true); 8249 effect->setEnabled(false); 8250 } 8251 } 8252 } else { 8253 if (index < 0) { 8254 return; 8255 } 8256 desc = mSuspendedEffects.valueAt(index); 8257 if (desc->mRefCount <= 0) { 8258 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8259 desc->mRefCount = 1; 8260 } 8261 if (--desc->mRefCount == 0) { 8262 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8263 if (desc->mEffect != 0) { 8264 sp<EffectModule> effect = desc->mEffect.promote(); 8265 if (effect != 0) { 8266 effect->setSuspended(false); 8267 sp<EffectHandle> handle = effect->controlHandle(); 8268 if (handle != 0) { 8269 effect->setEnabled(handle->enabled()); 8270 } 8271 } 8272 desc->mEffect.clear(); 8273 } 8274 mSuspendedEffects.removeItemsAt(index); 8275 } 8276 } 8277} 8278 8279// must be called with ThreadBase::mLock held 8280void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8281{ 8282 sp<SuspendedEffectDesc> desc; 8283 8284 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8285 if (suspend) { 8286 if (index >= 0) { 8287 desc = mSuspendedEffects.valueAt(index); 8288 } else { 8289 desc = new SuspendedEffectDesc(); 8290 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8291 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8292 } 8293 if (desc->mRefCount++ == 0) { 8294 Vector< sp<EffectModule> > effects; 8295 getSuspendEligibleEffects(effects); 8296 for (size_t i = 0; i < effects.size(); i++) { 8297 setEffectSuspended_l(&effects[i]->desc().type, true); 8298 } 8299 } 8300 } else { 8301 if (index < 0) { 8302 return; 8303 } 8304 desc = mSuspendedEffects.valueAt(index); 8305 if (desc->mRefCount <= 0) { 8306 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8307 desc->mRefCount = 1; 8308 } 8309 if (--desc->mRefCount == 0) { 8310 Vector<const effect_uuid_t *> types; 8311 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8312 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8313 continue; 8314 } 8315 types.add(&mSuspendedEffects.valueAt(i)->mType); 8316 } 8317 for (size_t i = 0; i < types.size(); i++) { 8318 setEffectSuspended_l(types[i], false); 8319 } 8320 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8321 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8322 } 8323 } 8324} 8325 8326 8327// The volume effect is used for automated tests only 8328#ifndef OPENSL_ES_H_ 8329static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8330 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8331const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8332#endif //OPENSL_ES_H_ 8333 8334bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8335{ 8336 // auxiliary effects and visualizer are never suspended on output mix 8337 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8338 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8339 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8340 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8341 return false; 8342 } 8343 return true; 8344} 8345 8346void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8347{ 8348 effects.clear(); 8349 for (size_t i = 0; i < mEffects.size(); i++) { 8350 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8351 effects.add(mEffects[i]); 8352 } 8353 } 8354} 8355 8356sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8357 const effect_uuid_t *type) 8358{ 8359 sp<EffectModule> effect = getEffectFromType_l(type); 8360 return effect != 0 && effect->isEnabled() ? effect : 0; 8361} 8362 8363void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8364 bool enabled) 8365{ 8366 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8367 if (enabled) { 8368 if (index < 0) { 8369 // if the effect is not suspend check if all effects are suspended 8370 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8371 if (index < 0) { 8372 return; 8373 } 8374 if (!isEffectEligibleForSuspend(effect->desc())) { 8375 return; 8376 } 8377 setEffectSuspended_l(&effect->desc().type, enabled); 8378 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8379 if (index < 0) { 8380 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8381 return; 8382 } 8383 } 8384 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8385 effect->desc().type.timeLow); 8386 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8387 // if effect is requested to suspended but was not yet enabled, supend it now. 8388 if (desc->mEffect == 0) { 8389 desc->mEffect = effect; 8390 effect->setEnabled(false); 8391 effect->setSuspended(true); 8392 } 8393 } else { 8394 if (index < 0) { 8395 return; 8396 } 8397 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8398 effect->desc().type.timeLow); 8399 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8400 desc->mEffect.clear(); 8401 effect->setSuspended(false); 8402 } 8403} 8404 8405#undef LOG_TAG 8406#define LOG_TAG "AudioFlinger" 8407 8408// ---------------------------------------------------------------------------- 8409 8410status_t AudioFlinger::onTransact( 8411 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8412{ 8413 return BnAudioFlinger::onTransact(code, data, reply, flags); 8414} 8415 8416}; // namespace android 8417