AudioFlinger.cpp revision 3e07470f3b122097cacfe5b85cdb1359279a2f33
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#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51#include "ServiceUtilities.h" 52 53#include <media/EffectsFactoryApi.h> 54#include <audio_effects/effect_visualizer.h> 55#include <audio_effects/effect_ns.h> 56#include <audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <cpustats/ThreadCpuUsage.h> 61#include <powermanager/PowerManager.h> 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63 64#include <common_time/cc_helper.h> 65#include <common_time/local_clock.h> 66 67// ---------------------------------------------------------------------------- 68 69 70namespace android { 71 72static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74 75static const float MAX_GAIN = 4096.0f; 76static const uint32_t MAX_GAIN_INT = 0x1000; 77 78// retry counts for buffer fill timeout 79// 50 * ~20msecs = 1 second 80static const int8_t kMaxTrackRetries = 50; 81static const int8_t kMaxTrackStartupRetries = 50; 82// allow less retry attempts on direct output thread. 83// direct outputs can be a scarce resource in audio hardware and should 84// be released as quickly as possible. 85static const int8_t kMaxTrackRetriesDirect = 2; 86 87static const int kDumpLockRetries = 50; 88static const int kDumpLockSleepUs = 20000; 89 90// don't warn about blocked writes or record buffer overflows more often than this 91static const nsecs_t kWarningThrottleNs = seconds(5); 92 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_IDLE; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_IDLE; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid cnt\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 // FIXME dead, remove from IAudioFlinger 436 uint32_t flags, 437 const sp<IMemory>& sharedBuffer, 438 audio_io_handle_t output, 439 bool isTimed, 440 int *sessionId, 441 status_t *status) 442{ 443 sp<PlaybackThread::Track> track; 444 sp<TrackHandle> trackHandle; 445 sp<Client> client; 446 status_t lStatus; 447 int lSessionId; 448 449 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 450 // but if someone uses binder directly they could bypass that and cause us to crash 451 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 452 ALOGE("createTrack() invalid stream type %d", streamType); 453 lStatus = BAD_VALUE; 454 goto Exit; 455 } 456 457 { 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 PlaybackThread *effectThread = NULL; 461 if (thread == NULL) { 462 ALOGE("unknown output thread"); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 client = registerPid_l(pid); 468 469 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 470 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 471 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 472 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 473 if (mPlaybackThreads.keyAt(i) != output) { 474 // prevent same audio session on different output threads 475 uint32_t sessions = t->hasAudioSession(*sessionId); 476 if (sessions & PlaybackThread::TRACK_SESSION) { 477 ALOGE("createTrack() session ID %d already in use", *sessionId); 478 lStatus = BAD_VALUE; 479 goto Exit; 480 } 481 // check if an effect with same session ID is waiting for a track to be created 482 if (sessions & PlaybackThread::EFFECT_SESSION) { 483 effectThread = t.get(); 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 } 508 if (lStatus == NO_ERROR) { 509 trackHandle = new TrackHandle(track); 510 } else { 511 // remove local strong reference to Client before deleting the Track so that the Client 512 // destructor is called by the TrackBase destructor with mLock held 513 client.clear(); 514 track.clear(); 515 } 516 517Exit: 518 if(status) { 519 *status = lStatus; 520 } 521 return trackHandle; 522} 523 524uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 525{ 526 Mutex::Autolock _l(mLock); 527 PlaybackThread *thread = checkPlaybackThread_l(output); 528 if (thread == NULL) { 529 ALOGW("sampleRate() unknown thread %d", output); 530 return 0; 531 } 532 return thread->sampleRate(); 533} 534 535int AudioFlinger::channelCount(audio_io_handle_t output) const 536{ 537 Mutex::Autolock _l(mLock); 538 PlaybackThread *thread = checkPlaybackThread_l(output); 539 if (thread == NULL) { 540 ALOGW("channelCount() unknown thread %d", output); 541 return 0; 542 } 543 return thread->channelCount(); 544} 545 546audio_format_t AudioFlinger::format(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("format() unknown thread %d", output); 552 return AUDIO_FORMAT_INVALID; 553 } 554 return thread->format(); 555} 556 557size_t AudioFlinger::frameCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("frameCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->frameCount(); 566} 567 568uint32_t AudioFlinger::latency(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("latency() unknown thread %d", output); 574 return 0; 575 } 576 return thread->latency(); 577} 578 579status_t AudioFlinger::setMasterVolume(float value) 580{ 581 status_t ret = initCheck(); 582 if (ret != NO_ERROR) { 583 return ret; 584 } 585 586 // check calling permissions 587 if (!settingsAllowed()) { 588 return PERMISSION_DENIED; 589 } 590 591 float swmv = value; 592 593 // when hw supports master volume, don't scale in sw mixer 594 if (MVS_NONE != mMasterVolumeSupportLvl) { 595 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 596 AutoMutex lock(mHardwareLock); 597 audio_hw_device_t *dev = mAudioHwDevs[i]; 598 599 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 600 if (NULL != dev->set_master_volume) { 601 dev->set_master_volume(dev, value); 602 } 603 mHardwareStatus = AUDIO_HW_IDLE; 604 } 605 606 swmv = 1.0; 607 } 608 609 Mutex::Autolock _l(mLock); 610 mMasterVolume = value; 611 mMasterVolumeSW = swmv; 612 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 613 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 614 615 return NO_ERROR; 616} 617 618status_t AudioFlinger::setMode(audio_mode_t mode) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 630 ALOGW("Illegal value: setMode(%d)", mode); 631 return BAD_VALUE; 632 } 633 634 { // scope for the lock 635 AutoMutex lock(mHardwareLock); 636 mHardwareStatus = AUDIO_HW_SET_MODE; 637 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 if (NO_ERROR == ret) { 642 Mutex::Autolock _l(mLock); 643 mMode = mode; 644 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 645 mPlaybackThreads.valueAt(i)->setMode(mode); 646 } 647 648 return ret; 649} 650 651status_t AudioFlinger::setMicMute(bool state) 652{ 653 status_t ret = initCheck(); 654 if (ret != NO_ERROR) { 655 return ret; 656 } 657 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 AutoMutex lock(mHardwareLock); 664 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 665 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 666 mHardwareStatus = AUDIO_HW_IDLE; 667 return ret; 668} 669 670bool AudioFlinger::getMicMute() const 671{ 672 status_t ret = initCheck(); 673 if (ret != NO_ERROR) { 674 return false; 675 } 676 677 bool state = AUDIO_MODE_INVALID; 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 680 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return state; 683} 684 685status_t AudioFlinger::setMasterMute(bool muted) 686{ 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 Mutex::Autolock _l(mLock); 693 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 694 mMasterMute = muted; 695 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 696 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 697 698 return NO_ERROR; 699} 700 701float AudioFlinger::masterVolume() const 702{ 703 Mutex::Autolock _l(mLock); 704 return masterVolume_l(); 705} 706 707float AudioFlinger::masterVolumeSW() const 708{ 709 Mutex::Autolock _l(mLock); 710 return masterVolumeSW_l(); 711} 712 713bool AudioFlinger::masterMute() const 714{ 715 Mutex::Autolock _l(mLock); 716 return masterMute_l(); 717} 718 719float AudioFlinger::masterVolume_l() const 720{ 721 if (MVS_FULL == mMasterVolumeSupportLvl) { 722 float ret_val; 723 AutoMutex lock(mHardwareLock); 724 725 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 726 assert(NULL != mPrimaryHardwareDev); 727 assert(NULL != mPrimaryHardwareDev->get_master_volume); 728 729 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 730 mHardwareStatus = AUDIO_HW_IDLE; 731 return ret_val; 732 } 733 734 return mMasterVolume; 735} 736 737status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 738 audio_io_handle_t output) 739{ 740 // check calling permissions 741 if (!settingsAllowed()) { 742 return PERMISSION_DENIED; 743 } 744 745 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 746 ALOGE("setStreamVolume() invalid stream %d", stream); 747 return BAD_VALUE; 748 } 749 750 AutoMutex lock(mLock); 751 PlaybackThread *thread = NULL; 752 if (output) { 753 thread = checkPlaybackThread_l(output); 754 if (thread == NULL) { 755 return BAD_VALUE; 756 } 757 } 758 759 mStreamTypes[stream].volume = value; 760 761 if (thread == NULL) { 762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 763 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 764 } 765 } else { 766 thread->setStreamVolume(stream, value); 767 } 768 769 return NO_ERROR; 770} 771 772status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 773{ 774 // check calling permissions 775 if (!settingsAllowed()) { 776 return PERMISSION_DENIED; 777 } 778 779 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 780 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 781 ALOGE("setStreamMute() invalid stream %d", stream); 782 return BAD_VALUE; 783 } 784 785 AutoMutex lock(mLock); 786 mStreamTypes[stream].mute = muted; 787 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 788 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 789 790 return NO_ERROR; 791} 792 793float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 794{ 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 796 return 0.0f; 797 } 798 799 AutoMutex lock(mLock); 800 float volume; 801 if (output) { 802 PlaybackThread *thread = checkPlaybackThread_l(output); 803 if (thread == NULL) { 804 return 0.0f; 805 } 806 volume = thread->streamVolume(stream); 807 } else { 808 volume = streamVolume_l(stream); 809 } 810 811 return volume; 812} 813 814bool AudioFlinger::streamMute(audio_stream_type_t stream) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return true; 818 } 819 820 AutoMutex lock(mLock); 821 return streamMute_l(stream); 822} 823 824status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 825{ 826 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 827 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 828 // check calling permissions 829 if (!settingsAllowed()) { 830 return PERMISSION_DENIED; 831 } 832 833 // ioHandle == 0 means the parameters are global to the audio hardware interface 834 if (ioHandle == 0) { 835 status_t final_result = NO_ERROR; 836 { 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 839 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 840 audio_hw_device_t *dev = mAudioHwDevs[i]; 841 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 842 final_result = result ?: final_result; 843 } 844 mHardwareStatus = AUDIO_HW_IDLE; 845 } 846 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 847 AudioParameter param = AudioParameter(keyValuePairs); 848 String8 value; 849 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 850 Mutex::Autolock _l(mLock); 851 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 852 if (mBtNrecIsOff != btNrecIsOff) { 853 for (size_t i = 0; i < mRecordThreads.size(); i++) { 854 sp<RecordThread> thread = mRecordThreads.valueAt(i); 855 RecordThread::RecordTrack *track = thread->track(); 856 if (track != NULL) { 857 audio_devices_t device = (audio_devices_t)( 858 thread->device() & AUDIO_DEVICE_IN_ALL); 859 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 860 thread->setEffectSuspended(FX_IID_AEC, 861 suspend, 862 track->sessionId()); 863 thread->setEffectSuspended(FX_IID_NS, 864 suspend, 865 track->sessionId()); 866 } 867 } 868 mBtNrecIsOff = btNrecIsOff; 869 } 870 } 871 return final_result; 872 } 873 874 // hold a strong ref on thread in case closeOutput() or closeInput() is called 875 // and the thread is exited once the lock is released 876 sp<ThreadBase> thread; 877 { 878 Mutex::Autolock _l(mLock); 879 thread = checkPlaybackThread_l(ioHandle); 880 if (thread == NULL) { 881 thread = checkRecordThread_l(ioHandle); 882 } else if (thread == primaryPlaybackThread_l()) { 883 // indicate output device change to all input threads for pre processing 884 AudioParameter param = AudioParameter(keyValuePairs); 885 int value; 886 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 887 for (size_t i = 0; i < mRecordThreads.size(); i++) { 888 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 889 } 890 } 891 } 892 } 893 if (thread != 0) { 894 return thread->setParameters(keyValuePairs); 895 } 896 return BAD_VALUE; 897} 898 899String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 900{ 901// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 902// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 903 904 if (ioHandle == 0) { 905 String8 out_s8; 906 907 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 908 char *s; 909 { 910 AutoMutex lock(mHardwareLock); 911 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 912 audio_hw_device_t *dev = mAudioHwDevs[i]; 913 s = dev->get_parameters(dev, keys.string()); 914 mHardwareStatus = AUDIO_HW_IDLE; 915 } 916 out_s8 += String8(s ? s : ""); 917 free(s); 918 } 919 return out_s8; 920 } 921 922 Mutex::Autolock _l(mLock); 923 924 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 925 if (playbackThread != NULL) { 926 return playbackThread->getParameters(keys); 927 } 928 RecordThread *recordThread = checkRecordThread_l(ioHandle); 929 if (recordThread != NULL) { 930 return recordThread->getParameters(keys); 931 } 932 return String8(""); 933} 934 935size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 936{ 937 status_t ret = initCheck(); 938 if (ret != NO_ERROR) { 939 return 0; 940 } 941 942 AutoMutex lock(mHardwareLock); 943 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 944 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 945 mHardwareStatus = AUDIO_HW_IDLE; 946 return size; 947} 948 949unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 950{ 951 if (ioHandle == 0) { 952 return 0; 953 } 954 955 Mutex::Autolock _l(mLock); 956 957 RecordThread *recordThread = checkRecordThread_l(ioHandle); 958 if (recordThread != NULL) { 959 return recordThread->getInputFramesLost(); 960 } 961 return 0; 962} 963 964status_t AudioFlinger::setVoiceVolume(float value) 965{ 966 status_t ret = initCheck(); 967 if (ret != NO_ERROR) { 968 return ret; 969 } 970 971 // check calling permissions 972 if (!settingsAllowed()) { 973 return PERMISSION_DENIED; 974 } 975 976 AutoMutex lock(mHardwareLock); 977 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 978 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 979 mHardwareStatus = AUDIO_HW_IDLE; 980 981 return ret; 982} 983 984status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 985 audio_io_handle_t output) const 986{ 987 status_t status; 988 989 Mutex::Autolock _l(mLock); 990 991 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 992 if (playbackThread != NULL) { 993 return playbackThread->getRenderPosition(halFrames, dspFrames); 994 } 995 996 return BAD_VALUE; 997} 998 999void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1000{ 1001 1002 Mutex::Autolock _l(mLock); 1003 1004 pid_t pid = IPCThreadState::self()->getCallingPid(); 1005 if (mNotificationClients.indexOfKey(pid) < 0) { 1006 sp<NotificationClient> notificationClient = new NotificationClient(this, 1007 client, 1008 pid); 1009 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1010 1011 mNotificationClients.add(pid, notificationClient); 1012 1013 sp<IBinder> binder = client->asBinder(); 1014 binder->linkToDeath(notificationClient); 1015 1016 // the config change is always sent from playback or record threads to avoid deadlock 1017 // with AudioSystem::gLock 1018 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1019 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1020 } 1021 1022 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1023 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1024 } 1025 } 1026} 1027 1028void AudioFlinger::removeNotificationClient(pid_t pid) 1029{ 1030 Mutex::Autolock _l(mLock); 1031 1032 mNotificationClients.removeItem(pid); 1033 1034 ALOGV("%d died, releasing its sessions", pid); 1035 size_t num = mAudioSessionRefs.size(); 1036 bool removed = false; 1037 for (size_t i = 0; i< num; ) { 1038 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1039 ALOGV(" pid %d @ %d", ref->pid, i); 1040 if (ref->pid == pid) { 1041 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1042 mAudioSessionRefs.removeAt(i); 1043 delete ref; 1044 removed = true; 1045 num--; 1046 } else { 1047 i++; 1048 } 1049 } 1050 if (removed) { 1051 purgeStaleEffects_l(); 1052 } 1053} 1054 1055// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1056void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1057{ 1058 size_t size = mNotificationClients.size(); 1059 for (size_t i = 0; i < size; i++) { 1060 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1061 param2); 1062 } 1063} 1064 1065// removeClient_l() must be called with AudioFlinger::mLock held 1066void AudioFlinger::removeClient_l(pid_t pid) 1067{ 1068 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1069 mClients.removeItem(pid); 1070} 1071 1072 1073// ---------------------------------------------------------------------------- 1074 1075AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1076 uint32_t device, type_t type) 1077 : Thread(false), 1078 mType(type), 1079 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1080 // mChannelMask 1081 mChannelCount(0), 1082 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1083 mParamStatus(NO_ERROR), 1084 mStandby(false), mId(id), 1085 mDevice(device), 1086 mDeathRecipient(new PMDeathRecipient(this)) 1087{ 1088} 1089 1090AudioFlinger::ThreadBase::~ThreadBase() 1091{ 1092 mParamCond.broadcast(); 1093 // do not lock the mutex in destructor 1094 releaseWakeLock_l(); 1095 if (mPowerManager != 0) { 1096 sp<IBinder> binder = mPowerManager->asBinder(); 1097 binder->unlinkToDeath(mDeathRecipient); 1098 } 1099} 1100 1101void AudioFlinger::ThreadBase::exit() 1102{ 1103 ALOGV("ThreadBase::exit"); 1104 { 1105 // This lock prevents the following race in thread (uniprocessor for illustration): 1106 // if (!exitPending()) { 1107 // // context switch from here to exit() 1108 // // exit() calls requestExit(), what exitPending() observes 1109 // // exit() calls signal(), which is dropped since no waiters 1110 // // context switch back from exit() to here 1111 // mWaitWorkCV.wait(...); 1112 // // now thread is hung 1113 // } 1114 AutoMutex lock(mLock); 1115 requestExit(); 1116 mWaitWorkCV.signal(); 1117 } 1118 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1119 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1120 requestExitAndWait(); 1121} 1122 1123status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1124{ 1125 status_t status; 1126 1127 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1128 Mutex::Autolock _l(mLock); 1129 1130 mNewParameters.add(keyValuePairs); 1131 mWaitWorkCV.signal(); 1132 // wait condition with timeout in case the thread loop has exited 1133 // before the request could be processed 1134 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1135 status = mParamStatus; 1136 mWaitWorkCV.signal(); 1137 } else { 1138 status = TIMED_OUT; 1139 } 1140 return status; 1141} 1142 1143void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1144{ 1145 Mutex::Autolock _l(mLock); 1146 sendConfigEvent_l(event, param); 1147} 1148 1149// sendConfigEvent_l() must be called with ThreadBase::mLock held 1150void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1151{ 1152 ConfigEvent configEvent; 1153 configEvent.mEvent = event; 1154 configEvent.mParam = param; 1155 mConfigEvents.add(configEvent); 1156 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1157 mWaitWorkCV.signal(); 1158} 1159 1160void AudioFlinger::ThreadBase::processConfigEvents() 1161{ 1162 mLock.lock(); 1163 while(!mConfigEvents.isEmpty()) { 1164 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1165 ConfigEvent configEvent = mConfigEvents[0]; 1166 mConfigEvents.removeAt(0); 1167 // release mLock before locking AudioFlinger mLock: lock order is always 1168 // AudioFlinger then ThreadBase to avoid cross deadlock 1169 mLock.unlock(); 1170 mAudioFlinger->mLock.lock(); 1171 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1172 mAudioFlinger->mLock.unlock(); 1173 mLock.lock(); 1174 } 1175 mLock.unlock(); 1176} 1177 1178status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1179{ 1180 const size_t SIZE = 256; 1181 char buffer[SIZE]; 1182 String8 result; 1183 1184 bool locked = tryLock(mLock); 1185 if (!locked) { 1186 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1187 write(fd, buffer, strlen(buffer)); 1188 } 1189 1190 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1191 result.append(buffer); 1192 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1193 result.append(buffer); 1194 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1195 result.append(buffer); 1196 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1197 result.append(buffer); 1198 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1199 result.append(buffer); 1200 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1201 result.append(buffer); 1202 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1203 result.append(buffer); 1204 1205 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1206 result.append(buffer); 1207 result.append(" Index Command"); 1208 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1209 snprintf(buffer, SIZE, "\n %02d ", i); 1210 result.append(buffer); 1211 result.append(mNewParameters[i]); 1212 } 1213 1214 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, " Index event param\n"); 1217 result.append(buffer); 1218 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1219 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1220 result.append(buffer); 1221 } 1222 result.append("\n"); 1223 1224 write(fd, result.string(), result.size()); 1225 1226 if (locked) { 1227 mLock.unlock(); 1228 } 1229 return NO_ERROR; 1230} 1231 1232status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1233{ 1234 const size_t SIZE = 256; 1235 char buffer[SIZE]; 1236 String8 result; 1237 1238 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1239 write(fd, buffer, strlen(buffer)); 1240 1241 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1242 sp<EffectChain> chain = mEffectChains[i]; 1243 if (chain != 0) { 1244 chain->dump(fd, args); 1245 } 1246 } 1247 return NO_ERROR; 1248} 1249 1250void AudioFlinger::ThreadBase::acquireWakeLock() 1251{ 1252 Mutex::Autolock _l(mLock); 1253 acquireWakeLock_l(); 1254} 1255 1256void AudioFlinger::ThreadBase::acquireWakeLock_l() 1257{ 1258 if (mPowerManager == 0) { 1259 // use checkService() to avoid blocking if power service is not up yet 1260 sp<IBinder> binder = 1261 defaultServiceManager()->checkService(String16("power")); 1262 if (binder == 0) { 1263 ALOGW("Thread %s cannot connect to the power manager service", mName); 1264 } else { 1265 mPowerManager = interface_cast<IPowerManager>(binder); 1266 binder->linkToDeath(mDeathRecipient); 1267 } 1268 } 1269 if (mPowerManager != 0) { 1270 sp<IBinder> binder = new BBinder(); 1271 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1272 binder, 1273 String16(mName)); 1274 if (status == NO_ERROR) { 1275 mWakeLockToken = binder; 1276 } 1277 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1278 } 1279} 1280 1281void AudioFlinger::ThreadBase::releaseWakeLock() 1282{ 1283 Mutex::Autolock _l(mLock); 1284 releaseWakeLock_l(); 1285} 1286 1287void AudioFlinger::ThreadBase::releaseWakeLock_l() 1288{ 1289 if (mWakeLockToken != 0) { 1290 ALOGV("releaseWakeLock_l() %s", mName); 1291 if (mPowerManager != 0) { 1292 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1293 } 1294 mWakeLockToken.clear(); 1295 } 1296} 1297 1298void AudioFlinger::ThreadBase::clearPowerManager() 1299{ 1300 Mutex::Autolock _l(mLock); 1301 releaseWakeLock_l(); 1302 mPowerManager.clear(); 1303} 1304 1305void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1306{ 1307 sp<ThreadBase> thread = mThread.promote(); 1308 if (thread != 0) { 1309 thread->clearPowerManager(); 1310 } 1311 ALOGW("power manager service died !!!"); 1312} 1313 1314void AudioFlinger::ThreadBase::setEffectSuspended( 1315 const effect_uuid_t *type, bool suspend, int sessionId) 1316{ 1317 Mutex::Autolock _l(mLock); 1318 setEffectSuspended_l(type, suspend, sessionId); 1319} 1320 1321void AudioFlinger::ThreadBase::setEffectSuspended_l( 1322 const effect_uuid_t *type, bool suspend, int sessionId) 1323{ 1324 sp<EffectChain> chain = getEffectChain_l(sessionId); 1325 if (chain != 0) { 1326 if (type != NULL) { 1327 chain->setEffectSuspended_l(type, suspend); 1328 } else { 1329 chain->setEffectSuspendedAll_l(suspend); 1330 } 1331 } 1332 1333 updateSuspendedSessions_l(type, suspend, sessionId); 1334} 1335 1336void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1337{ 1338 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1339 if (index < 0) { 1340 return; 1341 } 1342 1343 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1344 mSuspendedSessions.editValueAt(index); 1345 1346 for (size_t i = 0; i < sessionEffects.size(); i++) { 1347 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1348 for (int j = 0; j < desc->mRefCount; j++) { 1349 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1350 chain->setEffectSuspendedAll_l(true); 1351 } else { 1352 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1353 desc->mType.timeLow); 1354 chain->setEffectSuspended_l(&desc->mType, true); 1355 } 1356 } 1357 } 1358} 1359 1360void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1361 bool suspend, 1362 int sessionId) 1363{ 1364 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1365 1366 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1367 1368 if (suspend) { 1369 if (index >= 0) { 1370 sessionEffects = mSuspendedSessions.editValueAt(index); 1371 } else { 1372 mSuspendedSessions.add(sessionId, sessionEffects); 1373 } 1374 } else { 1375 if (index < 0) { 1376 return; 1377 } 1378 sessionEffects = mSuspendedSessions.editValueAt(index); 1379 } 1380 1381 1382 int key = EffectChain::kKeyForSuspendAll; 1383 if (type != NULL) { 1384 key = type->timeLow; 1385 } 1386 index = sessionEffects.indexOfKey(key); 1387 1388 sp <SuspendedSessionDesc> desc; 1389 if (suspend) { 1390 if (index >= 0) { 1391 desc = sessionEffects.valueAt(index); 1392 } else { 1393 desc = new SuspendedSessionDesc(); 1394 if (type != NULL) { 1395 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1396 } 1397 sessionEffects.add(key, desc); 1398 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1399 } 1400 desc->mRefCount++; 1401 } else { 1402 if (index < 0) { 1403 return; 1404 } 1405 desc = sessionEffects.valueAt(index); 1406 if (--desc->mRefCount == 0) { 1407 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1408 sessionEffects.removeItemsAt(index); 1409 if (sessionEffects.isEmpty()) { 1410 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1411 sessionId); 1412 mSuspendedSessions.removeItem(sessionId); 1413 } 1414 } 1415 } 1416 if (!sessionEffects.isEmpty()) { 1417 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1418 } 1419} 1420 1421void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1422 bool enabled, 1423 int sessionId) 1424{ 1425 Mutex::Autolock _l(mLock); 1426 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1427} 1428 1429void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1430 bool enabled, 1431 int sessionId) 1432{ 1433 if (mType != RECORD) { 1434 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1435 // another session. This gives the priority to well behaved effect control panels 1436 // and applications not using global effects. 1437 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1438 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1439 } 1440 } 1441 1442 sp<EffectChain> chain = getEffectChain_l(sessionId); 1443 if (chain != 0) { 1444 chain->checkSuspendOnEffectEnabled(effect, enabled); 1445 } 1446} 1447 1448// ---------------------------------------------------------------------------- 1449 1450AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1451 AudioStreamOut* output, 1452 audio_io_handle_t id, 1453 uint32_t device, 1454 type_t type) 1455 : ThreadBase(audioFlinger, id, device, type), 1456 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1457 // Assumes constructor is called by AudioFlinger with it's mLock held, 1458 // but it would be safer to explicitly pass initial masterMute as parameter 1459 mMasterMute(audioFlinger->masterMute_l()), 1460 // mStreamTypes[] initialized in constructor body 1461 mOutput(output), 1462 // Assumes constructor is called by AudioFlinger with it's mLock held, 1463 // but it would be safer to explicitly pass initial masterVolume as parameter 1464 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1465 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1466{ 1467 snprintf(mName, kNameLength, "AudioOut_%d", id); 1468 1469 readOutputParameters(); 1470 1471 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1472 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1473 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1474 stream = (audio_stream_type_t) (stream + 1)) { 1475 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1476 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1477 // initialized by stream_type_t default constructor 1478 // mStreamTypes[stream].valid = true; 1479 } 1480 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1481 // because mAudioFlinger doesn't have one to copy from 1482} 1483 1484AudioFlinger::PlaybackThread::~PlaybackThread() 1485{ 1486 delete [] mMixBuffer; 1487} 1488 1489status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1490{ 1491 dumpInternals(fd, args); 1492 dumpTracks(fd, args); 1493 dumpEffectChains(fd, args); 1494 return NO_ERROR; 1495} 1496 1497status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1498{ 1499 const size_t SIZE = 256; 1500 char buffer[SIZE]; 1501 String8 result; 1502 1503 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1504 result.append(buffer); 1505 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1506 for (size_t i = 0; i < mTracks.size(); ++i) { 1507 sp<Track> track = mTracks[i]; 1508 if (track != 0) { 1509 track->dump(buffer, SIZE); 1510 result.append(buffer); 1511 } 1512 } 1513 1514 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1515 result.append(buffer); 1516 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1517 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1518 sp<Track> track = mActiveTracks[i].promote(); 1519 if (track != 0) { 1520 track->dump(buffer, SIZE); 1521 result.append(buffer); 1522 } 1523 } 1524 write(fd, result.string(), result.size()); 1525 return NO_ERROR; 1526} 1527 1528status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1529{ 1530 const size_t SIZE = 256; 1531 char buffer[SIZE]; 1532 String8 result; 1533 1534 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1535 result.append(buffer); 1536 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1537 result.append(buffer); 1538 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1539 result.append(buffer); 1540 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1541 result.append(buffer); 1542 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1543 result.append(buffer); 1544 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1545 result.append(buffer); 1546 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1547 result.append(buffer); 1548 write(fd, result.string(), result.size()); 1549 1550 dumpBase(fd, args); 1551 1552 return NO_ERROR; 1553} 1554 1555// Thread virtuals 1556status_t AudioFlinger::PlaybackThread::readyToRun() 1557{ 1558 status_t status = initCheck(); 1559 if (status == NO_ERROR) { 1560 ALOGI("AudioFlinger's thread %p ready to run", this); 1561 } else { 1562 ALOGE("No working audio driver found."); 1563 } 1564 return status; 1565} 1566 1567void AudioFlinger::PlaybackThread::onFirstRef() 1568{ 1569 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1570} 1571 1572// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1573sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1574 const sp<AudioFlinger::Client>& client, 1575 audio_stream_type_t streamType, 1576 uint32_t sampleRate, 1577 audio_format_t format, 1578 uint32_t channelMask, 1579 int frameCount, 1580 const sp<IMemory>& sharedBuffer, 1581 int sessionId, 1582 bool isTimed, 1583 status_t *status) 1584{ 1585 sp<Track> track; 1586 status_t lStatus; 1587 1588 if (mType == DIRECT) { 1589 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1590 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1591 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1592 "for output %p with format %d", 1593 sampleRate, format, channelMask, mOutput, mFormat); 1594 lStatus = BAD_VALUE; 1595 goto Exit; 1596 } 1597 } 1598 } else { 1599 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1600 if (sampleRate > mSampleRate*2) { 1601 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1602 lStatus = BAD_VALUE; 1603 goto Exit; 1604 } 1605 } 1606 1607 lStatus = initCheck(); 1608 if (lStatus != NO_ERROR) { 1609 ALOGE("Audio driver not initialized."); 1610 goto Exit; 1611 } 1612 1613 { // scope for mLock 1614 Mutex::Autolock _l(mLock); 1615 1616 // all tracks in same audio session must share the same routing strategy otherwise 1617 // conflicts will happen when tracks are moved from one output to another by audio policy 1618 // manager 1619 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1620 for (size_t i = 0; i < mTracks.size(); ++i) { 1621 sp<Track> t = mTracks[i]; 1622 if (t != 0) { 1623 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1624 if (sessionId == t->sessionId() && strategy != actual) { 1625 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1626 strategy, actual); 1627 lStatus = BAD_VALUE; 1628 goto Exit; 1629 } 1630 } 1631 } 1632 1633 if (!isTimed) { 1634 track = new Track(this, client, streamType, sampleRate, format, 1635 channelMask, frameCount, sharedBuffer, sessionId); 1636 } else { 1637 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1638 channelMask, frameCount, sharedBuffer, sessionId); 1639 } 1640 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1641 lStatus = NO_MEMORY; 1642 goto Exit; 1643 } 1644 mTracks.add(track); 1645 1646 sp<EffectChain> chain = getEffectChain_l(sessionId); 1647 if (chain != 0) { 1648 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1649 track->setMainBuffer(chain->inBuffer()); 1650 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1651 chain->incTrackCnt(); 1652 } 1653 1654 // invalidate track immediately if the stream type was moved to another thread since 1655 // createTrack() was called by the client process. 1656 if (!mStreamTypes[streamType].valid) { 1657 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1658 this, streamType); 1659 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1660 } 1661 } 1662 lStatus = NO_ERROR; 1663 1664Exit: 1665 if(status) { 1666 *status = lStatus; 1667 } 1668 return track; 1669} 1670 1671uint32_t AudioFlinger::PlaybackThread::latency() const 1672{ 1673 Mutex::Autolock _l(mLock); 1674 if (initCheck() == NO_ERROR) { 1675 return mOutput->stream->get_latency(mOutput->stream); 1676 } else { 1677 return 0; 1678 } 1679} 1680 1681void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1682{ 1683 Mutex::Autolock _l(mLock); 1684 mMasterVolume = value; 1685} 1686 1687void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1688{ 1689 Mutex::Autolock _l(mLock); 1690 setMasterMute_l(muted); 1691} 1692 1693void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1694{ 1695 Mutex::Autolock _l(mLock); 1696 mStreamTypes[stream].volume = value; 1697} 1698 1699void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1700{ 1701 Mutex::Autolock _l(mLock); 1702 mStreamTypes[stream].mute = muted; 1703} 1704 1705float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1706{ 1707 Mutex::Autolock _l(mLock); 1708 return mStreamTypes[stream].volume; 1709} 1710 1711// addTrack_l() must be called with ThreadBase::mLock held 1712status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1713{ 1714 status_t status = ALREADY_EXISTS; 1715 1716 // set retry count for buffer fill 1717 track->mRetryCount = kMaxTrackStartupRetries; 1718 if (mActiveTracks.indexOf(track) < 0) { 1719 // the track is newly added, make sure it fills up all its 1720 // buffers before playing. This is to ensure the client will 1721 // effectively get the latency it requested. 1722 track->mFillingUpStatus = Track::FS_FILLING; 1723 track->mResetDone = false; 1724 mActiveTracks.add(track); 1725 if (track->mainBuffer() != mMixBuffer) { 1726 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1727 if (chain != 0) { 1728 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1729 chain->incActiveTrackCnt(); 1730 } 1731 } 1732 1733 status = NO_ERROR; 1734 } 1735 1736 ALOGV("mWaitWorkCV.broadcast"); 1737 mWaitWorkCV.broadcast(); 1738 1739 return status; 1740} 1741 1742// destroyTrack_l() must be called with ThreadBase::mLock held 1743void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1744{ 1745 track->mState = TrackBase::TERMINATED; 1746 if (mActiveTracks.indexOf(track) < 0) { 1747 removeTrack_l(track); 1748 } 1749} 1750 1751void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1752{ 1753 mTracks.remove(track); 1754 deleteTrackName_l(track->name()); 1755 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1756 if (chain != 0) { 1757 chain->decTrackCnt(); 1758 } 1759} 1760 1761String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1762{ 1763 String8 out_s8 = String8(""); 1764 char *s; 1765 1766 Mutex::Autolock _l(mLock); 1767 if (initCheck() != NO_ERROR) { 1768 return out_s8; 1769 } 1770 1771 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1772 out_s8 = String8(s); 1773 free(s); 1774 return out_s8; 1775} 1776 1777// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1778void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1779 AudioSystem::OutputDescriptor desc; 1780 void *param2 = NULL; 1781 1782 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1783 1784 switch (event) { 1785 case AudioSystem::OUTPUT_OPENED: 1786 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1787 desc.channels = mChannelMask; 1788 desc.samplingRate = mSampleRate; 1789 desc.format = mFormat; 1790 desc.frameCount = mFrameCount; 1791 desc.latency = latency(); 1792 param2 = &desc; 1793 break; 1794 1795 case AudioSystem::STREAM_CONFIG_CHANGED: 1796 param2 = ¶m; 1797 case AudioSystem::OUTPUT_CLOSED: 1798 default: 1799 break; 1800 } 1801 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1802} 1803 1804void AudioFlinger::PlaybackThread::readOutputParameters() 1805{ 1806 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1807 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1808 mChannelCount = (uint16_t)popcount(mChannelMask); 1809 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1810 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1811 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1812 1813 // FIXME - Current mixer implementation only supports stereo output: Always 1814 // Allocate a stereo buffer even if HW output is mono. 1815 delete[] mMixBuffer; 1816 mMixBuffer = new int16_t[mFrameCount * 2]; 1817 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1818 1819 // force reconfiguration of effect chains and engines to take new buffer size and audio 1820 // parameters into account 1821 // Note that mLock is not held when readOutputParameters() is called from the constructor 1822 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1823 // matter. 1824 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1825 Vector< sp<EffectChain> > effectChains = mEffectChains; 1826 for (size_t i = 0; i < effectChains.size(); i ++) { 1827 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1828 } 1829} 1830 1831status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1832{ 1833 if (halFrames == NULL || dspFrames == NULL) { 1834 return BAD_VALUE; 1835 } 1836 Mutex::Autolock _l(mLock); 1837 if (initCheck() != NO_ERROR) { 1838 return INVALID_OPERATION; 1839 } 1840 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1841 1842 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1843} 1844 1845uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1846{ 1847 Mutex::Autolock _l(mLock); 1848 uint32_t result = 0; 1849 if (getEffectChain_l(sessionId) != 0) { 1850 result = EFFECT_SESSION; 1851 } 1852 1853 for (size_t i = 0; i < mTracks.size(); ++i) { 1854 sp<Track> track = mTracks[i]; 1855 if (sessionId == track->sessionId() && 1856 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1857 result |= TRACK_SESSION; 1858 break; 1859 } 1860 } 1861 1862 return result; 1863} 1864 1865uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1866{ 1867 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1868 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1869 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1870 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1871 } 1872 for (size_t i = 0; i < mTracks.size(); i++) { 1873 sp<Track> track = mTracks[i]; 1874 if (sessionId == track->sessionId() && 1875 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1876 return AudioSystem::getStrategyForStream(track->streamType()); 1877 } 1878 } 1879 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1880} 1881 1882 1883AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1884{ 1885 Mutex::Autolock _l(mLock); 1886 return mOutput; 1887} 1888 1889AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1890{ 1891 Mutex::Autolock _l(mLock); 1892 AudioStreamOut *output = mOutput; 1893 mOutput = NULL; 1894 return output; 1895} 1896 1897// this method must always be called either with ThreadBase mLock held or inside the thread loop 1898audio_stream_t* AudioFlinger::PlaybackThread::stream() 1899{ 1900 if (mOutput == NULL) { 1901 return NULL; 1902 } 1903 return &mOutput->stream->common; 1904} 1905 1906uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1907{ 1908 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1909 // decoding and transfer time. So sleeping for half of the latency would likely cause 1910 // underruns 1911 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1912 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1913 } else { 1914 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1915 } 1916} 1917 1918// ---------------------------------------------------------------------------- 1919 1920AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1921 audio_io_handle_t id, uint32_t device, type_t type) 1922 : PlaybackThread(audioFlinger, output, id, device, type), 1923 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1924 mPrevMixerStatus(MIXER_IDLE) 1925{ 1926 // FIXME - Current mixer implementation only supports stereo output 1927 if (mChannelCount == 1) { 1928 ALOGE("Invalid audio hardware channel count"); 1929 } 1930} 1931 1932AudioFlinger::MixerThread::~MixerThread() 1933{ 1934 delete mAudioMixer; 1935} 1936 1937class CpuStats { 1938public: 1939 void sample(); 1940#ifdef DEBUG_CPU_USAGE 1941private: 1942 ThreadCpuUsage mCpu; 1943#endif 1944}; 1945 1946void CpuStats::sample() { 1947#ifdef DEBUG_CPU_USAGE 1948 const CentralTendencyStatistics& stats = mCpu.statistics(); 1949 mCpu.sampleAndEnable(); 1950 unsigned n = stats.n(); 1951 // mCpu.elapsed() is expensive, so don't call it every loop 1952 if ((n & 127) == 1) { 1953 long long elapsed = mCpu.elapsed(); 1954 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1955 double perLoop = elapsed / (double) n; 1956 double perLoop100 = perLoop * 0.01; 1957 double mean = stats.mean(); 1958 double stddev = stats.stddev(); 1959 double minimum = stats.minimum(); 1960 double maximum = stats.maximum(); 1961 mCpu.resetStatistics(); 1962 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1963 elapsed * .000000001, n, perLoop * .000001, 1964 mean * .001, 1965 stddev * .001, 1966 minimum * .001, 1967 maximum * .001, 1968 mean / perLoop100, 1969 stddev / perLoop100, 1970 minimum / perLoop100, 1971 maximum / perLoop100); 1972 } 1973 } 1974#endif 1975}; 1976 1977void AudioFlinger::PlaybackThread::checkSilentMode_l() 1978{ 1979 if (!mMasterMute) { 1980 char value[PROPERTY_VALUE_MAX]; 1981 if (property_get("ro.audio.silent", value, "0") > 0) { 1982 char *endptr; 1983 unsigned long ul = strtoul(value, &endptr, 0); 1984 if (*endptr == '\0' && ul != 0) { 1985 ALOGD("Silence is golden"); 1986 // The setprop command will not allow a property to be changed after 1987 // the first time it is set, so we don't have to worry about un-muting. 1988 setMasterMute_l(true); 1989 } 1990 } 1991 } 1992} 1993 1994bool AudioFlinger::MixerThread::threadLoop() 1995{ 1996 // DirectOutputThread has single trackToRemove instead of Vector 1997 Vector< sp<Track> > tracksToRemove; 1998 // DirectOutputThread has activeTrack here 1999 nsecs_t standbyTime = systemTime(); 2000 size_t mixBufferSize = mFrameCount * mFrameSize; 2001 2002 // FIXME: Relaxed timing because of a certain device that can't meet latency 2003 // Should be reduced to 2x after the vendor fixes the driver issue 2004 // increase threshold again due to low power audio mode. The way this warning threshold is 2005 // calculated and its usefulness should be reconsidered anyway. 2006 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2007 nsecs_t lastWarning = 0; 2008 bool longStandbyExit = false; 2009 2010 uint32_t activeSleepTime = activeSleepTimeUs(); 2011 uint32_t idleSleepTime = idleSleepTimeUs(); 2012 uint32_t sleepTime = idleSleepTime; 2013 2014 uint32_t sleepTimeShift = 0; 2015 CpuStats cpuStats; 2016 2017 // DirectOutputThread has shorter standbyDelay 2018 2019 acquireWakeLock(); 2020 2021 while (!exitPending()) 2022 { 2023 cpuStats.sample(); 2024 2025 // DirectOutputThread has rampVolume, leftVol, rightVol 2026 2027 Vector< sp<EffectChain> > effectChains; 2028 2029 processConfigEvents(); 2030 2031 mixer_state mixerStatus = MIXER_IDLE; 2032 { // scope for mLock 2033 2034 Mutex::Autolock _l(mLock); 2035 2036 if (checkForNewParameters_l()) { 2037 mixBufferSize = mFrameCount * mFrameSize; 2038 2039 // FIXME: Relaxed timing because of a certain device that can't meet latency 2040 // Should be reduced to 2x after the vendor fixes the driver issue 2041 // increase threshold again due to low power audio mode. The way this warning 2042 // threshold is calculated and its usefulness should be reconsidered anyway. 2043 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2044 2045 activeSleepTime = activeSleepTimeUs(); 2046 idleSleepTime = idleSleepTimeUs(); 2047 // DirectOutputThread updates standbyDelay also 2048 } 2049 2050 // put audio hardware into standby after short delay 2051 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2052 mSuspended > 0)) { 2053 if (!mStandby) { 2054 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2055 mOutput->stream->common.standby(&mOutput->stream->common); 2056 mStandby = true; 2057 mBytesWritten = 0; 2058 } 2059 2060 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2061 // we're about to wait, flush the binder command buffer 2062 IPCThreadState::self()->flushCommands(); 2063 2064 if (exitPending()) break; 2065 2066 releaseWakeLock_l(); 2067 // wait until we have something to do... 2068 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2069 mWaitWorkCV.wait(mLock); 2070 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2071 acquireWakeLock_l(); 2072 2073 mPrevMixerStatus = MIXER_IDLE; 2074 checkSilentMode_l(); 2075 2076 standbyTime = systemTime() + mStandbyTimeInNsecs; 2077 sleepTime = idleSleepTime; 2078 sleepTimeShift = 0; 2079 continue; 2080 } 2081 } 2082 2083 mixerStatus = prepareTracks_l(&tracksToRemove); 2084 2085 // prevent any changes in effect chain list and in each effect chain 2086 // during mixing and effect process as the audio buffers could be deleted 2087 // or modified if an effect is created or deleted 2088 lockEffectChains_l(effectChains); 2089 } 2090 2091 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2092 // obtain the presentation timestamp of the next output buffer 2093 int64_t pts; 2094 status_t status = INVALID_OPERATION; 2095 2096 if (NULL != mOutput->stream->get_next_write_timestamp) { 2097 status = mOutput->stream->get_next_write_timestamp( 2098 mOutput->stream, &pts); 2099 } 2100 2101 if (status != NO_ERROR) { 2102 pts = AudioBufferProvider::kInvalidPTS; 2103 } 2104 2105 // mix buffers... 2106 mAudioMixer->process(pts); 2107 // increase sleep time progressively when application underrun condition clears. 2108 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2109 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2110 // such that we would underrun the audio HAL. 2111 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2112 sleepTimeShift--; 2113 } 2114 sleepTime = 0; 2115 standbyTime = systemTime() + mStandbyTimeInNsecs; 2116 //TODO: delay standby when effects have a tail 2117 } else { 2118 // If no tracks are ready, sleep once for the duration of an output 2119 // buffer size, then write 0s to the output 2120 if (sleepTime == 0) { 2121 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2122 sleepTime = activeSleepTime >> sleepTimeShift; 2123 if (sleepTime < kMinThreadSleepTimeUs) { 2124 sleepTime = kMinThreadSleepTimeUs; 2125 } 2126 // reduce sleep time in case of consecutive application underruns to avoid 2127 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2128 // duration we would end up writing less data than needed by the audio HAL if 2129 // the condition persists. 2130 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2131 sleepTimeShift++; 2132 } 2133 } else { 2134 sleepTime = idleSleepTime; 2135 } 2136 } else if (mBytesWritten != 0 || 2137 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2138 memset (mMixBuffer, 0, mixBufferSize); 2139 sleepTime = 0; 2140 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2141 } 2142 // TODO add standby time extension fct of effect tail 2143 } 2144 2145 if (mSuspended > 0) { 2146 sleepTime = suspendSleepTimeUs(); 2147 } 2148 2149 // only process effects if we're going to write 2150 if (sleepTime == 0) { 2151 2152 // DirectOutputThread adds applyVolume here 2153 2154 for (size_t i = 0; i < effectChains.size(); i ++) { 2155 effectChains[i]->process_l(); 2156 } 2157 } 2158 2159 // enable changes in effect chain 2160 unlockEffectChains(effectChains); 2161 2162 // sleepTime == 0 means we must write to audio hardware 2163 if (sleepTime == 0) { 2164 // FIXME Only in MixerThread, and rewrite to reduce number of system calls 2165 mLastWriteTime = systemTime(); 2166 mInWrite = true; 2167 mBytesWritten += mixBufferSize; 2168 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2169 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2170 mNumWrites++; 2171 mInWrite = false; 2172 2173 // Only in MixerThread: start of write blocked detection 2174 nsecs_t now = systemTime(); 2175 nsecs_t delta = now - mLastWriteTime; 2176 if (!mStandby && delta > maxPeriod) { 2177 mNumDelayedWrites++; 2178 if ((now - lastWarning) > kWarningThrottleNs) { 2179 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2180 ns2ms(delta), mNumDelayedWrites, this); 2181 lastWarning = now; 2182 } 2183 if (mStandby) { 2184 longStandbyExit = true; 2185 } 2186 } 2187 // end of write blocked detection 2188 2189 mStandby = false; 2190 } else { 2191 usleep(sleepTime); 2192 } 2193 2194 // finally let go of removed track(s), without the lock held 2195 // since we can't guarantee the destructors won't acquire that 2196 // same lock. 2197 tracksToRemove.clear(); 2198 2199 // Effect chains will be actually deleted here if they were removed from 2200 // mEffectChains list during mixing or effects processing 2201 effectChains.clear(); 2202 2203 // FIXME Note that the above .clear() is no longer necessary since effectChains 2204 // is now local to this block, but will keep it for now (at least until merge done). 2205 } 2206 2207 // put output stream into standby mode 2208 if (!mStandby) { 2209 mOutput->stream->common.standby(&mOutput->stream->common); 2210 } 2211 2212 releaseWakeLock(); 2213 2214 ALOGV("Thread %p type %d exiting", this, mType); 2215 return false; 2216} 2217 2218// prepareTracks_l() must be called with ThreadBase::mLock held 2219AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2220 Vector< sp<Track> > *tracksToRemove) 2221{ 2222 2223 mixer_state mixerStatus = MIXER_IDLE; 2224 // find out which tracks need to be processed 2225 size_t count = mActiveTracks.size(); 2226 size_t mixedTracks = 0; 2227 size_t tracksWithEffect = 0; 2228 2229 float masterVolume = mMasterVolume; 2230 bool masterMute = mMasterMute; 2231 2232 if (masterMute) { 2233 masterVolume = 0; 2234 } 2235 // Delegate master volume control to effect in output mix effect chain if needed 2236 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2237 if (chain != 0) { 2238 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2239 chain->setVolume_l(&v, &v); 2240 masterVolume = (float)((v + (1 << 23)) >> 24); 2241 chain.clear(); 2242 } 2243 2244 for (size_t i=0 ; i<count ; i++) { 2245 sp<Track> t = mActiveTracks[i].promote(); 2246 if (t == 0) continue; 2247 2248 // this const just means the local variable doesn't change 2249 Track* const track = t.get(); 2250 audio_track_cblk_t* cblk = track->cblk(); 2251 2252 // The first time a track is added we wait 2253 // for all its buffers to be filled before processing it 2254 int name = track->name(); 2255 // make sure that we have enough frames to mix one full buffer. 2256 // enforce this condition only once to enable draining the buffer in case the client 2257 // app does not call stop() and relies on underrun to stop: 2258 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2259 // during last round 2260 uint32_t minFrames = 1; 2261 if (!track->isStopped() && !track->isPausing() && 2262 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2263 if (t->sampleRate() == (int)mSampleRate) { 2264 minFrames = mFrameCount; 2265 } else { 2266 // +1 for rounding and +1 for additional sample needed for interpolation 2267 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2268 // add frames already consumed but not yet released by the resampler 2269 // because cblk->framesReady() will include these frames 2270 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2271 // the minimum track buffer size is normally twice the number of frames necessary 2272 // to fill one buffer and the resampler should not leave more than one buffer worth 2273 // of unreleased frames after each pass, but just in case... 2274 ALOG_ASSERT(minFrames <= cblk->frameCount); 2275 } 2276 } 2277 if ((track->framesReady() >= minFrames) && track->isReady() && 2278 !track->isPaused() && !track->isTerminated()) 2279 { 2280 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2281 2282 mixedTracks++; 2283 2284 // track->mainBuffer() != mMixBuffer means there is an effect chain 2285 // connected to the track 2286 chain.clear(); 2287 if (track->mainBuffer() != mMixBuffer) { 2288 chain = getEffectChain_l(track->sessionId()); 2289 // Delegate volume control to effect in track effect chain if needed 2290 if (chain != 0) { 2291 tracksWithEffect++; 2292 } else { 2293 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2294 name, track->sessionId()); 2295 } 2296 } 2297 2298 2299 int param = AudioMixer::VOLUME; 2300 if (track->mFillingUpStatus == Track::FS_FILLED) { 2301 // no ramp for the first volume setting 2302 track->mFillingUpStatus = Track::FS_ACTIVE; 2303 if (track->mState == TrackBase::RESUMING) { 2304 track->mState = TrackBase::ACTIVE; 2305 param = AudioMixer::RAMP_VOLUME; 2306 } 2307 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2308 } else if (cblk->server != 0) { 2309 // If the track is stopped before the first frame was mixed, 2310 // do not apply ramp 2311 param = AudioMixer::RAMP_VOLUME; 2312 } 2313 2314 // compute volume for this track 2315 uint32_t vl, vr, va; 2316 if (track->isMuted() || track->isPausing() || 2317 mStreamTypes[track->streamType()].mute) { 2318 vl = vr = va = 0; 2319 if (track->isPausing()) { 2320 track->setPaused(); 2321 } 2322 } else { 2323 2324 // read original volumes with volume control 2325 float typeVolume = mStreamTypes[track->streamType()].volume; 2326 float v = masterVolume * typeVolume; 2327 uint32_t vlr = cblk->getVolumeLR(); 2328 vl = vlr & 0xFFFF; 2329 vr = vlr >> 16; 2330 // track volumes come from shared memory, so can't be trusted and must be clamped 2331 if (vl > MAX_GAIN_INT) { 2332 ALOGV("Track left volume out of range: %04X", vl); 2333 vl = MAX_GAIN_INT; 2334 } 2335 if (vr > MAX_GAIN_INT) { 2336 ALOGV("Track right volume out of range: %04X", vr); 2337 vr = MAX_GAIN_INT; 2338 } 2339 // now apply the master volume and stream type volume 2340 vl = (uint32_t)(v * vl) << 12; 2341 vr = (uint32_t)(v * vr) << 12; 2342 // assuming master volume and stream type volume each go up to 1.0, 2343 // vl and vr are now in 8.24 format 2344 2345 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2346 // send level comes from shared memory and so may be corrupt 2347 if (sendLevel > MAX_GAIN_INT) { 2348 ALOGV("Track send level out of range: %04X", sendLevel); 2349 sendLevel = MAX_GAIN_INT; 2350 } 2351 va = (uint32_t)(v * sendLevel); 2352 } 2353 // Delegate volume control to effect in track effect chain if needed 2354 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2355 // Do not ramp volume if volume is controlled by effect 2356 param = AudioMixer::VOLUME; 2357 track->mHasVolumeController = true; 2358 } else { 2359 // force no volume ramp when volume controller was just disabled or removed 2360 // from effect chain to avoid volume spike 2361 if (track->mHasVolumeController) { 2362 param = AudioMixer::VOLUME; 2363 } 2364 track->mHasVolumeController = false; 2365 } 2366 2367 // Convert volumes from 8.24 to 4.12 format 2368 // This additional clamping is needed in case chain->setVolume_l() overshot 2369 vl = (vl + (1 << 11)) >> 12; 2370 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2371 vr = (vr + (1 << 11)) >> 12; 2372 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2373 2374 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2375 2376 // XXX: these things DON'T need to be done each time 2377 mAudioMixer->setBufferProvider(name, track); 2378 mAudioMixer->enable(name); 2379 2380 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2381 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2382 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2383 mAudioMixer->setParameter( 2384 name, 2385 AudioMixer::TRACK, 2386 AudioMixer::FORMAT, (void *)track->format()); 2387 mAudioMixer->setParameter( 2388 name, 2389 AudioMixer::TRACK, 2390 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2391 mAudioMixer->setParameter( 2392 name, 2393 AudioMixer::RESAMPLE, 2394 AudioMixer::SAMPLE_RATE, 2395 (void *)(cblk->sampleRate)); 2396 mAudioMixer->setParameter( 2397 name, 2398 AudioMixer::TRACK, 2399 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2400 mAudioMixer->setParameter( 2401 name, 2402 AudioMixer::TRACK, 2403 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2404 2405 // reset retry count 2406 track->mRetryCount = kMaxTrackRetries; 2407 // If one track is ready, set the mixer ready if: 2408 // - the mixer was not ready during previous round OR 2409 // - no other track is not ready 2410 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2411 mixerStatus != MIXER_TRACKS_ENABLED) { 2412 mixerStatus = MIXER_TRACKS_READY; 2413 } 2414 } else { 2415 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2416 if (track->isStopped()) { 2417 track->reset(); 2418 } 2419 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2420 // We have consumed all the buffers of this track. 2421 // Remove it from the list of active tracks. 2422 tracksToRemove->add(track); 2423 } else { 2424 // No buffers for this track. Give it a few chances to 2425 // fill a buffer, then remove it from active list. 2426 if (--(track->mRetryCount) <= 0) { 2427 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2428 tracksToRemove->add(track); 2429 // indicate to client process that the track was disabled because of underrun 2430 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2431 // If one track is not ready, mark the mixer also not ready if: 2432 // - the mixer was ready during previous round OR 2433 // - no other track is ready 2434 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2435 mixerStatus != MIXER_TRACKS_READY) { 2436 mixerStatus = MIXER_TRACKS_ENABLED; 2437 } 2438 } 2439 mAudioMixer->disable(name); 2440 } 2441 } 2442 2443 // remove all the tracks that need to be... 2444 count = tracksToRemove->size(); 2445 if (CC_UNLIKELY(count)) { 2446 for (size_t i=0 ; i<count ; i++) { 2447 const sp<Track>& track = tracksToRemove->itemAt(i); 2448 mActiveTracks.remove(track); 2449 if (track->mainBuffer() != mMixBuffer) { 2450 chain = getEffectChain_l(track->sessionId()); 2451 if (chain != 0) { 2452 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2453 chain->decActiveTrackCnt(); 2454 } 2455 } 2456 if (track->isTerminated()) { 2457 removeTrack_l(track); 2458 } 2459 } 2460 } 2461 2462 // mix buffer must be cleared if all tracks are connected to an 2463 // effect chain as in this case the mixer will not write to 2464 // mix buffer and track effects will accumulate into it 2465 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2466 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2467 } 2468 2469 mPrevMixerStatus = mixerStatus; 2470 return mixerStatus; 2471} 2472 2473void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2474{ 2475 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2476 this, streamType, mTracks.size()); 2477 Mutex::Autolock _l(mLock); 2478 2479 size_t size = mTracks.size(); 2480 for (size_t i = 0; i < size; i++) { 2481 sp<Track> t = mTracks[i]; 2482 if (t->streamType() == streamType) { 2483 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2484 t->mCblk->cv.signal(); 2485 } 2486 } 2487} 2488 2489void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2490{ 2491 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2492 this, streamType, valid); 2493 Mutex::Autolock _l(mLock); 2494 2495 mStreamTypes[streamType].valid = valid; 2496} 2497 2498// getTrackName_l() must be called with ThreadBase::mLock held 2499int AudioFlinger::MixerThread::getTrackName_l() 2500{ 2501 return mAudioMixer->getTrackName(); 2502} 2503 2504// deleteTrackName_l() must be called with ThreadBase::mLock held 2505void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2506{ 2507 ALOGV("remove track (%d) and delete from mixer", name); 2508 mAudioMixer->deleteTrackName(name); 2509} 2510 2511// checkForNewParameters_l() must be called with ThreadBase::mLock held 2512bool AudioFlinger::MixerThread::checkForNewParameters_l() 2513{ 2514 bool reconfig = false; 2515 2516 while (!mNewParameters.isEmpty()) { 2517 status_t status = NO_ERROR; 2518 String8 keyValuePair = mNewParameters[0]; 2519 AudioParameter param = AudioParameter(keyValuePair); 2520 int value; 2521 2522 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2523 reconfig = true; 2524 } 2525 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2526 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2527 status = BAD_VALUE; 2528 } else { 2529 reconfig = true; 2530 } 2531 } 2532 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2533 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2534 status = BAD_VALUE; 2535 } else { 2536 reconfig = true; 2537 } 2538 } 2539 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2540 // do not accept frame count changes if tracks are open as the track buffer 2541 // size depends on frame count and correct behavior would not be guaranteed 2542 // if frame count is changed after track creation 2543 if (!mTracks.isEmpty()) { 2544 status = INVALID_OPERATION; 2545 } else { 2546 reconfig = true; 2547 } 2548 } 2549 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2550 // when changing the audio output device, call addBatteryData to notify 2551 // the change 2552 if ((int)mDevice != value) { 2553 uint32_t params = 0; 2554 // check whether speaker is on 2555 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2556 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2557 } 2558 2559 int deviceWithoutSpeaker 2560 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2561 // check if any other device (except speaker) is on 2562 if (value & deviceWithoutSpeaker ) { 2563 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2564 } 2565 2566 if (params != 0) { 2567 addBatteryData(params); 2568 } 2569 } 2570 2571 // forward device change to effects that have requested to be 2572 // aware of attached audio device. 2573 mDevice = (uint32_t)value; 2574 for (size_t i = 0; i < mEffectChains.size(); i++) { 2575 mEffectChains[i]->setDevice_l(mDevice); 2576 } 2577 } 2578 2579 if (status == NO_ERROR) { 2580 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2581 keyValuePair.string()); 2582 if (!mStandby && status == INVALID_OPERATION) { 2583 mOutput->stream->common.standby(&mOutput->stream->common); 2584 mStandby = true; 2585 mBytesWritten = 0; 2586 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2587 keyValuePair.string()); 2588 } 2589 if (status == NO_ERROR && reconfig) { 2590 delete mAudioMixer; 2591 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2592 mAudioMixer = NULL; 2593 readOutputParameters(); 2594 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2595 for (size_t i = 0; i < mTracks.size() ; i++) { 2596 int name = getTrackName_l(); 2597 if (name < 0) break; 2598 mTracks[i]->mName = name; 2599 // limit track sample rate to 2 x new output sample rate 2600 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2601 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2602 } 2603 } 2604 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2605 } 2606 } 2607 2608 mNewParameters.removeAt(0); 2609 2610 mParamStatus = status; 2611 mParamCond.signal(); 2612 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2613 // already timed out waiting for the status and will never signal the condition. 2614 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2615 } 2616 return reconfig; 2617} 2618 2619status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2620{ 2621 const size_t SIZE = 256; 2622 char buffer[SIZE]; 2623 String8 result; 2624 2625 PlaybackThread::dumpInternals(fd, args); 2626 2627 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2628 result.append(buffer); 2629 write(fd, result.string(), result.size()); 2630 return NO_ERROR; 2631} 2632 2633uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2634{ 2635 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2636} 2637 2638uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2639{ 2640 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2641} 2642 2643// ---------------------------------------------------------------------------- 2644AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2645 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2646 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2647 // mLeftVolFloat, mRightVolFloat 2648 // mLeftVolShort, mRightVolShort 2649{ 2650} 2651 2652AudioFlinger::DirectOutputThread::~DirectOutputThread() 2653{ 2654} 2655 2656void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2657{ 2658 // Do not apply volume on compressed audio 2659 if (!audio_is_linear_pcm(mFormat)) { 2660 return; 2661 } 2662 2663 // convert to signed 16 bit before volume calculation 2664 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2665 size_t count = mFrameCount * mChannelCount; 2666 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2667 int16_t *dst = mMixBuffer + count-1; 2668 while(count--) { 2669 *dst-- = (int16_t)(*src--^0x80) << 8; 2670 } 2671 } 2672 2673 size_t frameCount = mFrameCount; 2674 int16_t *out = mMixBuffer; 2675 if (ramp) { 2676 if (mChannelCount == 1) { 2677 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2678 int32_t vlInc = d / (int32_t)frameCount; 2679 int32_t vl = ((int32_t)mLeftVolShort << 16); 2680 do { 2681 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2682 out++; 2683 vl += vlInc; 2684 } while (--frameCount); 2685 2686 } else { 2687 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2688 int32_t vlInc = d / (int32_t)frameCount; 2689 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2690 int32_t vrInc = d / (int32_t)frameCount; 2691 int32_t vl = ((int32_t)mLeftVolShort << 16); 2692 int32_t vr = ((int32_t)mRightVolShort << 16); 2693 do { 2694 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2695 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2696 out += 2; 2697 vl += vlInc; 2698 vr += vrInc; 2699 } while (--frameCount); 2700 } 2701 } else { 2702 if (mChannelCount == 1) { 2703 do { 2704 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2705 out++; 2706 } while (--frameCount); 2707 } else { 2708 do { 2709 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2710 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2711 out += 2; 2712 } while (--frameCount); 2713 } 2714 } 2715 2716 // convert back to unsigned 8 bit after volume calculation 2717 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2718 size_t count = mFrameCount * mChannelCount; 2719 int16_t *src = mMixBuffer; 2720 uint8_t *dst = (uint8_t *)mMixBuffer; 2721 while(count--) { 2722 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2723 } 2724 } 2725 2726 mLeftVolShort = leftVol; 2727 mRightVolShort = rightVol; 2728} 2729 2730bool AudioFlinger::DirectOutputThread::threadLoop() 2731{ 2732 // MixerThread has Vector instead of single trackToRemove 2733 sp<Track> trackToRemove; 2734 2735 nsecs_t standbyTime = systemTime(); 2736 size_t mixBufferSize = mFrameCount * mFrameSize; 2737 2738 // MixerThread has relaxed timing: maxPeriod, lastWarning, longStandbyExit 2739 2740 uint32_t activeSleepTime = activeSleepTimeUs(); 2741 uint32_t idleSleepTime = idleSleepTimeUs(); 2742 uint32_t sleepTime = idleSleepTime; 2743 2744 // MixerThread has sleepTimeShift and cpuStats 2745 2746 // use shorter standby delay as on normal output to release 2747 // hardware resources as soon as possible 2748 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2749 2750 acquireWakeLock(); 2751 2752 while (!exitPending()) 2753 { 2754 // MixerThread has cpuStats.sample() 2755 2756 bool rampVolume; 2757 uint16_t leftVol; 2758 uint16_t rightVol; 2759 2760 Vector< sp<EffectChain> > effectChains; 2761 2762 processConfigEvents(); 2763 2764 // MixerThread does not have activeTrack here 2765 sp<Track> activeTrack; 2766 2767 mixer_state mixerStatus = MIXER_IDLE; 2768 { // scope for the mLock 2769 2770 Mutex::Autolock _l(mLock); 2771 2772 if (checkForNewParameters_l()) { 2773 mixBufferSize = mFrameCount * mFrameSize; 2774 2775 // different calculations here 2776 standbyDelay = microseconds(activeSleepTime*2); 2777 2778 activeSleepTime = activeSleepTimeUs(); 2779 idleSleepTime = idleSleepTimeUs(); 2780 standbyDelay = microseconds(activeSleepTime*2); 2781 } 2782 2783 // put audio hardware into standby after short delay 2784 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2785 mSuspended > 0)) { 2786 if (!mStandby) { 2787 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2788 mOutput->stream->common.standby(&mOutput->stream->common); 2789 mStandby = true; 2790 mBytesWritten = 0; 2791 } 2792 2793 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2794 // we're about to wait, flush the binder command buffer 2795 IPCThreadState::self()->flushCommands(); 2796 2797 if (exitPending()) break; 2798 2799 releaseWakeLock_l(); 2800 // wait until we have something to do... 2801 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2802 mWaitWorkCV.wait(mLock); 2803 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2804 acquireWakeLock_l(); 2805 2806 // MixerThread has "mPrevMixerStatus = MIXER_IDLE" 2807 checkSilentMode_l(); 2808 2809 // MixerThread has different standbyDelay 2810 standbyTime = systemTime() + standbyDelay; 2811 sleepTime = idleSleepTime; 2812 // MixerThread has "sleepTimeShift = 0" 2813 continue; 2814 } 2815 } 2816 2817 // MixerThread has "mixerStatus = prepareTracks_l(...)" 2818 2819 // equivalent to MixerThread's lockEffectChains_l, but without the lock 2820 // FIXME - is it OK to omit the lock here? 2821 effectChains = mEffectChains; 2822 2823 // find out which tracks need to be processed 2824 if (mActiveTracks.size() != 0) { 2825 sp<Track> t = mActiveTracks[0].promote(); 2826 if (t == 0) continue; 2827 2828 Track* const track = t.get(); 2829 audio_track_cblk_t* cblk = track->cblk(); 2830 2831 // The first time a track is added we wait 2832 // for all its buffers to be filled before processing it 2833 if (cblk->framesReady() && track->isReady() && 2834 !track->isPaused() && !track->isTerminated()) 2835 { 2836 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2837 2838 if (track->mFillingUpStatus == Track::FS_FILLED) { 2839 track->mFillingUpStatus = Track::FS_ACTIVE; 2840 mLeftVolFloat = mRightVolFloat = 0; 2841 mLeftVolShort = mRightVolShort = 0; 2842 if (track->mState == TrackBase::RESUMING) { 2843 track->mState = TrackBase::ACTIVE; 2844 rampVolume = true; 2845 } 2846 } else if (cblk->server != 0) { 2847 // If the track is stopped before the first frame was mixed, 2848 // do not apply ramp 2849 rampVolume = true; 2850 } 2851 // compute volume for this track 2852 float left, right; 2853 if (track->isMuted() || mMasterMute || track->isPausing() || 2854 mStreamTypes[track->streamType()].mute) { 2855 left = right = 0; 2856 if (track->isPausing()) { 2857 track->setPaused(); 2858 } 2859 } else { 2860 float typeVolume = mStreamTypes[track->streamType()].volume; 2861 float v = mMasterVolume * typeVolume; 2862 uint32_t vlr = cblk->getVolumeLR(); 2863 float v_clamped = v * (vlr & 0xFFFF); 2864 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2865 left = v_clamped/MAX_GAIN; 2866 v_clamped = v * (vlr >> 16); 2867 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2868 right = v_clamped/MAX_GAIN; 2869 } 2870 2871 if (left != mLeftVolFloat || right != mRightVolFloat) { 2872 mLeftVolFloat = left; 2873 mRightVolFloat = right; 2874 2875 // If audio HAL implements volume control, 2876 // force software volume to nominal value 2877 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2878 left = 1.0f; 2879 right = 1.0f; 2880 } 2881 2882 // Convert volumes from float to 8.24 2883 uint32_t vl = (uint32_t)(left * (1 << 24)); 2884 uint32_t vr = (uint32_t)(right * (1 << 24)); 2885 2886 // Delegate volume control to effect in track effect chain if needed 2887 // only one effect chain can be present on DirectOutputThread, so if 2888 // there is one, the track is connected to it 2889 if (!effectChains.isEmpty()) { 2890 // Do not ramp volume if volume is controlled by effect 2891 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2892 rampVolume = false; 2893 } 2894 } 2895 2896 // Convert volumes from 8.24 to 4.12 format 2897 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2898 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2899 leftVol = (uint16_t)v_clamped; 2900 v_clamped = (vr + (1 << 11)) >> 12; 2901 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2902 rightVol = (uint16_t)v_clamped; 2903 } else { 2904 leftVol = mLeftVolShort; 2905 rightVol = mRightVolShort; 2906 rampVolume = false; 2907 } 2908 2909 // reset retry count 2910 track->mRetryCount = kMaxTrackRetriesDirect; 2911 activeTrack = t; 2912 mixerStatus = MIXER_TRACKS_READY; 2913 } else { 2914 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2915 if (track->isStopped()) { 2916 track->reset(); 2917 } 2918 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2919 // We have consumed all the buffers of this track. 2920 // Remove it from the list of active tracks. 2921 trackToRemove = track; 2922 } else { 2923 // No buffers for this track. Give it a few chances to 2924 // fill a buffer, then remove it from active list. 2925 if (--(track->mRetryCount) <= 0) { 2926 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2927 trackToRemove = track; 2928 } else { 2929 mixerStatus = MIXER_TRACKS_ENABLED; 2930 } 2931 } 2932 } 2933 } 2934 2935 // remove all the tracks that need to be... 2936 if (CC_UNLIKELY(trackToRemove != 0)) { 2937 mActiveTracks.remove(trackToRemove); 2938 if (!effectChains.isEmpty()) { 2939 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2940 trackToRemove->sessionId()); 2941 effectChains[0]->decActiveTrackCnt(); 2942 } 2943 if (trackToRemove->isTerminated()) { 2944 removeTrack_l(trackToRemove); 2945 } 2946 } 2947 2948 lockEffectChains_l(effectChains); 2949 } 2950 2951 // For DirectOutputThread, this test is equivalent to "activeTrack != 0" 2952 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2953 AudioBufferProvider::Buffer buffer; 2954 size_t frameCount = mFrameCount; 2955 int8_t *curBuf = (int8_t *)mMixBuffer; 2956 // output audio to hardware 2957 while (frameCount) { 2958 buffer.frameCount = frameCount; 2959 activeTrack->getNextBuffer(&buffer); 2960 if (CC_UNLIKELY(buffer.raw == NULL)) { 2961 memset(curBuf, 0, frameCount * mFrameSize); 2962 break; 2963 } 2964 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2965 frameCount -= buffer.frameCount; 2966 curBuf += buffer.frameCount * mFrameSize; 2967 activeTrack->releaseBuffer(&buffer); 2968 } 2969 sleepTime = 0; 2970 standbyTime = systemTime() + standbyDelay; 2971 } else { 2972 if (sleepTime == 0) { 2973 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2974 sleepTime = activeSleepTime; 2975 } else { 2976 sleepTime = idleSleepTime; 2977 } 2978 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2979 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2980 sleepTime = 0; 2981 } 2982 } 2983 2984 if (mSuspended > 0) { 2985 sleepTime = suspendSleepTimeUs(); 2986 } 2987 2988 // only process effects if we're going to write 2989 if (sleepTime == 0) { 2990 2991 // MixerThread does not have applyVolume 2992 if (mixerStatus == MIXER_TRACKS_READY) { 2993 applyVolume(leftVol, rightVol, rampVolume); 2994 } 2995 2996 for (size_t i = 0; i < effectChains.size(); i ++) { 2997 effectChains[i]->process_l(); 2998 } 2999 } 3000 3001 // enable changes in effect chain 3002 unlockEffectChains(effectChains); 3003 3004 // sleepTime == 0 means we must write to audio hardware 3005 if (sleepTime == 0) { 3006 mLastWriteTime = systemTime(); 3007 mInWrite = true; 3008 mBytesWritten += mixBufferSize; 3009 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 3010 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 3011 mNumWrites++; 3012 mInWrite = false; 3013 3014 // MixerThread has write blocked detection here 3015 3016 mStandby = false; 3017 } else { 3018 usleep(sleepTime); 3019 } 3020 3021 // finally let go of removed track(s), without the lock held 3022 // since we can't guarantee the destructors won't acquire that 3023 // same lock. 3024 trackToRemove.clear(); 3025 activeTrack.clear(); 3026 3027 // Effect chains will be actually deleted here if they were removed from 3028 // mEffectChains list during mixing or effects processing 3029 effectChains.clear(); 3030 3031 // FIXME Note that the above .clear() is no longer necessary since effectChains 3032 // is now local to this block, but will keep it for now (at least until merge done). 3033 } 3034 3035 // put output stream into standby mode 3036 if (!mStandby) { 3037 mOutput->stream->common.standby(&mOutput->stream->common); 3038 } 3039 3040 releaseWakeLock(); 3041 3042 ALOGV("Thread %p type %d exiting", this, mType); 3043 return false; 3044} 3045 3046// getTrackName_l() must be called with ThreadBase::mLock held 3047int AudioFlinger::DirectOutputThread::getTrackName_l() 3048{ 3049 return 0; 3050} 3051 3052// deleteTrackName_l() must be called with ThreadBase::mLock held 3053void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3054{ 3055} 3056 3057// checkForNewParameters_l() must be called with ThreadBase::mLock held 3058bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3059{ 3060 bool reconfig = false; 3061 3062 while (!mNewParameters.isEmpty()) { 3063 status_t status = NO_ERROR; 3064 String8 keyValuePair = mNewParameters[0]; 3065 AudioParameter param = AudioParameter(keyValuePair); 3066 int value; 3067 3068 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3069 // do not accept frame count changes if tracks are open as the track buffer 3070 // size depends on frame count and correct behavior would not be garantied 3071 // if frame count is changed after track creation 3072 if (!mTracks.isEmpty()) { 3073 status = INVALID_OPERATION; 3074 } else { 3075 reconfig = true; 3076 } 3077 } 3078 if (status == NO_ERROR) { 3079 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3080 keyValuePair.string()); 3081 if (!mStandby && status == INVALID_OPERATION) { 3082 mOutput->stream->common.standby(&mOutput->stream->common); 3083 mStandby = true; 3084 mBytesWritten = 0; 3085 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3086 keyValuePair.string()); 3087 } 3088 if (status == NO_ERROR && reconfig) { 3089 readOutputParameters(); 3090 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3091 } 3092 } 3093 3094 mNewParameters.removeAt(0); 3095 3096 mParamStatus = status; 3097 mParamCond.signal(); 3098 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3099 // already timed out waiting for the status and will never signal the condition. 3100 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3101 } 3102 return reconfig; 3103} 3104 3105uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3106{ 3107 uint32_t time; 3108 if (audio_is_linear_pcm(mFormat)) { 3109 time = PlaybackThread::activeSleepTimeUs(); 3110 } else { 3111 time = 10000; 3112 } 3113 return time; 3114} 3115 3116uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3117{ 3118 uint32_t time; 3119 if (audio_is_linear_pcm(mFormat)) { 3120 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3121 } else { 3122 time = 10000; 3123 } 3124 return time; 3125} 3126 3127uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3128{ 3129 uint32_t time; 3130 if (audio_is_linear_pcm(mFormat)) { 3131 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3132 } else { 3133 time = 10000; 3134 } 3135 return time; 3136} 3137 3138 3139// ---------------------------------------------------------------------------- 3140 3141AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3142 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3143 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3144 mWaitTimeMs(UINT_MAX) 3145{ 3146 addOutputTrack(mainThread); 3147} 3148 3149AudioFlinger::DuplicatingThread::~DuplicatingThread() 3150{ 3151 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3152 mOutputTracks[i]->destroy(); 3153 } 3154} 3155 3156bool AudioFlinger::DuplicatingThread::threadLoop() 3157{ 3158 Vector< sp<Track> > tracksToRemove; 3159 nsecs_t standbyTime = systemTime(); 3160 size_t mixBufferSize = mFrameCount * mFrameSize; 3161 3162 // Only in DuplicatingThread 3163 SortedVector< sp<OutputTrack> > outputTracks; 3164 uint32_t writeFrames = 0; 3165 3166 uint32_t activeSleepTime = activeSleepTimeUs(); 3167 uint32_t idleSleepTime = idleSleepTimeUs(); 3168 uint32_t sleepTime = idleSleepTime; 3169 3170 acquireWakeLock(); 3171 3172 while (!exitPending()) 3173 { 3174 // MixerThread has cpuStats.sample 3175 3176 Vector< sp<EffectChain> > effectChains; 3177 3178 processConfigEvents(); 3179 3180 mixer_state mixerStatus = MIXER_IDLE; 3181 { // scope for the mLock 3182 3183 Mutex::Autolock _l(mLock); 3184 3185 if (checkForNewParameters_l()) { 3186 mixBufferSize = mFrameCount * mFrameSize; 3187 3188 // Only in DuplicatingThread 3189 updateWaitTime(); 3190 3191 activeSleepTime = activeSleepTimeUs(); 3192 idleSleepTime = idleSleepTimeUs(); 3193 } 3194 3195 // Only in DuplicatingThread 3196 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3197 outputTracks.add(mOutputTracks[i]); 3198 } 3199 3200 // put audio hardware into standby after short delay 3201 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 3202 mSuspended > 0)) { 3203 if (!mStandby) { 3204 // DuplicatingThread implements standby by stopping all tracks 3205 for (size_t i = 0; i < outputTracks.size(); i++) { 3206 outputTracks[i]->stop(); 3207 } 3208 mStandby = true; 3209 mBytesWritten = 0; 3210 } 3211 3212 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 3213 // we're about to wait, flush the binder command buffer 3214 IPCThreadState::self()->flushCommands(); 3215 outputTracks.clear(); 3216 3217 if (exitPending()) break; 3218 3219 releaseWakeLock_l(); 3220 // wait until we have something to do... 3221 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 3222 mWaitWorkCV.wait(mLock); 3223 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 3224 acquireWakeLock_l(); 3225 3226 // MixerThread has "mPrevMixerStatus = MIXER_IDLE" 3227 checkSilentMode_l(); 3228 3229 standbyTime = systemTime() + mStandbyTimeInNsecs; 3230 sleepTime = idleSleepTime; 3231 // MixerThread has sleepTimeShift 3232 continue; 3233 } 3234 } 3235 3236 mixerStatus = prepareTracks_l(&tracksToRemove); 3237 3238 // prevent any changes in effect chain list and in each effect chain 3239 // during mixing and effect process as the audio buffers could be deleted 3240 // or modified if an effect is created or deleted 3241 lockEffectChains_l(effectChains); 3242 } 3243 3244 // Duplicating Thread is completely different here 3245 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3246 // mix buffers... 3247 if (outputsReady(outputTracks)) { 3248 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3249 } else { 3250 memset(mMixBuffer, 0, mixBufferSize); 3251 } 3252 sleepTime = 0; 3253 writeFrames = mFrameCount; 3254 } else { 3255 if (sleepTime == 0) { 3256 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3257 sleepTime = activeSleepTime; 3258 } else { 3259 sleepTime = idleSleepTime; 3260 } 3261 } else if (mBytesWritten != 0) { 3262 // flush remaining overflow buffers in output tracks 3263 for (size_t i = 0; i < outputTracks.size(); i++) { 3264 if (outputTracks[i]->isActive()) { 3265 sleepTime = 0; 3266 writeFrames = 0; 3267 memset(mMixBuffer, 0, mixBufferSize); 3268 break; 3269 } 3270 } 3271 } 3272 } 3273 3274 if (mSuspended > 0) { 3275 sleepTime = suspendSleepTimeUs(); 3276 } 3277 3278 // only process effects if we're going to write 3279 if (sleepTime == 0) { 3280 for (size_t i = 0; i < effectChains.size(); i ++) { 3281 effectChains[i]->process_l(); 3282 } 3283 } 3284 3285 // enable changes in effect chain 3286 unlockEffectChains(effectChains); 3287 3288 // sleepTime == 0 means we must write to audio hardware 3289 if (sleepTime == 0) { 3290 standbyTime = systemTime() + mStandbyTimeInNsecs; 3291 for (size_t i = 0; i < outputTracks.size(); i++) { 3292 outputTracks[i]->write(mMixBuffer, writeFrames); 3293 } 3294 mStandby = false; 3295 mBytesWritten += mixBufferSize; 3296 3297 // MixerThread has write blocked detection here 3298 3299 } else { 3300 usleep(sleepTime); 3301 } 3302 3303 // finally let go of removed track(s), without the lock held 3304 // since we can't guarantee the destructors won't acquire that 3305 // same lock. 3306 tracksToRemove.clear(); 3307 outputTracks.clear(); 3308 3309 // Effect chains will be actually deleted here if they were removed from 3310 // mEffectChains list during mixing or effects processing 3311 effectChains.clear(); 3312 3313 // FIXME Note that the above .clear() is no longer necessary since effectChains 3314 // is now local to this block, but will keep it for now (at least until merge done). 3315 } 3316 3317 // MixerThread and DirectOutpuThread have standby here, 3318 // but for DuplicatingThread this is handled by the outputTracks 3319 3320 releaseWakeLock(); 3321 3322 ALOGV("Thread %p type %d exiting", this, mType); 3323 return false; 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(); 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(); 3353 return; 3354 } 3355 } 3356 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3357} 3358 3359void AudioFlinger::DuplicatingThread::updateWaitTime() 3360{ 3361 mWaitTimeMs = UINT_MAX; 3362 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3363 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3364 if (strong != 0) { 3365 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3366 if (waitTimeMs < mWaitTimeMs) { 3367 mWaitTimeMs = waitTimeMs; 3368 } 3369 } 3370 } 3371} 3372 3373 3374bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3375{ 3376 for (size_t i = 0; i < outputTracks.size(); i++) { 3377 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3378 if (thread == 0) { 3379 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3380 return false; 3381 } 3382 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3383 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3384 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3385 return false; 3386 } 3387 } 3388 return true; 3389} 3390 3391uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3392{ 3393 return (mWaitTimeMs * 1000) / 2; 3394} 3395 3396// ---------------------------------------------------------------------------- 3397 3398// TrackBase constructor must be called with AudioFlinger::mLock held 3399AudioFlinger::ThreadBase::TrackBase::TrackBase( 3400 ThreadBase *thread, 3401 const sp<Client>& client, 3402 uint32_t sampleRate, 3403 audio_format_t format, 3404 uint32_t channelMask, 3405 int frameCount, 3406 const sp<IMemory>& sharedBuffer, 3407 int sessionId) 3408 : RefBase(), 3409 mThread(thread), 3410 mClient(client), 3411 mCblk(NULL), 3412 // mBuffer 3413 // mBufferEnd 3414 mFrameCount(0), 3415 mState(IDLE), 3416 mFormat(format), 3417 mStepServerFailed(false), 3418 mSessionId(sessionId) 3419 // mChannelCount 3420 // mChannelMask 3421{ 3422 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3423 3424 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3425 size_t size = sizeof(audio_track_cblk_t); 3426 uint8_t channelCount = popcount(channelMask); 3427 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3428 if (sharedBuffer == 0) { 3429 size += bufferSize; 3430 } 3431 3432 if (client != NULL) { 3433 mCblkMemory = client->heap()->allocate(size); 3434 if (mCblkMemory != 0) { 3435 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3436 if (mCblk != NULL) { // construct the shared structure in-place. 3437 new(mCblk) audio_track_cblk_t(); 3438 // clear all buffers 3439 mCblk->frameCount = frameCount; 3440 mCblk->sampleRate = sampleRate; 3441 mChannelCount = channelCount; 3442 mChannelMask = channelMask; 3443 if (sharedBuffer == 0) { 3444 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3445 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3446 // Force underrun condition to avoid false underrun callback until first data is 3447 // written to buffer (other flags are cleared) 3448 mCblk->flags = CBLK_UNDERRUN_ON; 3449 } else { 3450 mBuffer = sharedBuffer->pointer(); 3451 } 3452 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3453 } 3454 } else { 3455 ALOGE("not enough memory for AudioTrack size=%u", size); 3456 client->heap()->dump("AudioTrack"); 3457 return; 3458 } 3459 } else { 3460 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3461 // construct the shared structure in-place. 3462 new(mCblk) audio_track_cblk_t(); 3463 // clear all buffers 3464 mCblk->frameCount = frameCount; 3465 mCblk->sampleRate = sampleRate; 3466 mChannelCount = channelCount; 3467 mChannelMask = channelMask; 3468 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3469 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3470 // Force underrun condition to avoid false underrun callback until first data is 3471 // written to buffer (other flags are cleared) 3472 mCblk->flags = CBLK_UNDERRUN_ON; 3473 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3474 } 3475} 3476 3477AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3478{ 3479 if (mCblk != NULL) { 3480 if (mClient == 0) { 3481 delete mCblk; 3482 } else { 3483 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3484 } 3485 } 3486 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3487 if (mClient != 0) { 3488 // Client destructor must run with AudioFlinger mutex locked 3489 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3490 // If the client's reference count drops to zero, the associated destructor 3491 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3492 // relying on the automatic clear() at end of scope. 3493 mClient.clear(); 3494 } 3495} 3496 3497// AudioBufferProvider interface 3498// getNextBuffer() = 0; 3499// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3500void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3501{ 3502 buffer->raw = NULL; 3503 mFrameCount = buffer->frameCount; 3504 (void) step(); // ignore return value of step() 3505 buffer->frameCount = 0; 3506} 3507 3508bool AudioFlinger::ThreadBase::TrackBase::step() { 3509 bool result; 3510 audio_track_cblk_t* cblk = this->cblk(); 3511 3512 result = cblk->stepServer(mFrameCount); 3513 if (!result) { 3514 ALOGV("stepServer failed acquiring cblk mutex"); 3515 mStepServerFailed = true; 3516 } 3517 return result; 3518} 3519 3520void AudioFlinger::ThreadBase::TrackBase::reset() { 3521 audio_track_cblk_t* cblk = this->cblk(); 3522 3523 cblk->user = 0; 3524 cblk->server = 0; 3525 cblk->userBase = 0; 3526 cblk->serverBase = 0; 3527 mStepServerFailed = false; 3528 ALOGV("TrackBase::reset"); 3529} 3530 3531int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3532 return (int)mCblk->sampleRate; 3533} 3534 3535void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3536 audio_track_cblk_t* cblk = this->cblk(); 3537 size_t frameSize = cblk->frameSize; 3538 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3539 int8_t *bufferEnd = bufferStart + frames * frameSize; 3540 3541 // Check validity of returned pointer in case the track control block would have been corrupted. 3542 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3543 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3544 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3545 server %d, serverBase %d, user %d, userBase %d", 3546 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3547 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3548 return NULL; 3549 } 3550 3551 return bufferStart; 3552} 3553 3554// ---------------------------------------------------------------------------- 3555 3556// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3557AudioFlinger::PlaybackThread::Track::Track( 3558 PlaybackThread *thread, 3559 const sp<Client>& client, 3560 audio_stream_type_t streamType, 3561 uint32_t sampleRate, 3562 audio_format_t format, 3563 uint32_t channelMask, 3564 int frameCount, 3565 const sp<IMemory>& sharedBuffer, 3566 int sessionId) 3567 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3568 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3569 mAuxEffectId(0), mHasVolumeController(false) 3570{ 3571 if (mCblk != NULL) { 3572 if (thread != NULL) { 3573 mName = thread->getTrackName_l(); 3574 mMainBuffer = thread->mixBuffer(); 3575 } 3576 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3577 if (mName < 0) { 3578 ALOGE("no more track names available"); 3579 } 3580 mStreamType = streamType; 3581 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3582 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3583 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3584 } 3585} 3586 3587AudioFlinger::PlaybackThread::Track::~Track() 3588{ 3589 ALOGV("PlaybackThread::Track destructor"); 3590 sp<ThreadBase> thread = mThread.promote(); 3591 if (thread != 0) { 3592 Mutex::Autolock _l(thread->mLock); 3593 mState = TERMINATED; 3594 } 3595} 3596 3597void AudioFlinger::PlaybackThread::Track::destroy() 3598{ 3599 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3600 // by removing it from mTracks vector, so there is a risk that this Tracks's 3601 // destructor is called. As the destructor needs to lock mLock, 3602 // we must acquire a strong reference on this Track before locking mLock 3603 // here so that the destructor is called only when exiting this function. 3604 // On the other hand, as long as Track::destroy() is only called by 3605 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3606 // this Track with its member mTrack. 3607 sp<Track> keep(this); 3608 { // scope for mLock 3609 sp<ThreadBase> thread = mThread.promote(); 3610 if (thread != 0) { 3611 if (!isOutputTrack()) { 3612 if (mState == ACTIVE || mState == RESUMING) { 3613 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3614 3615 // to track the speaker usage 3616 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3617 } 3618 AudioSystem::releaseOutput(thread->id()); 3619 } 3620 Mutex::Autolock _l(thread->mLock); 3621 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3622 playbackThread->destroyTrack_l(this); 3623 } 3624 } 3625} 3626 3627void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3628{ 3629 uint32_t vlr = mCblk->getVolumeLR(); 3630 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", 3631 mName - AudioMixer::TRACK0, 3632 (mClient == 0) ? getpid_cached : mClient->pid(), 3633 mStreamType, 3634 mFormat, 3635 mChannelMask, 3636 mSessionId, 3637 mFrameCount, 3638 mState, 3639 mMute, 3640 mFillingUpStatus, 3641 mCblk->sampleRate, 3642 vlr & 0xFFFF, 3643 vlr >> 16, 3644 mCblk->server, 3645 mCblk->user, 3646 (int)mMainBuffer, 3647 (int)mAuxBuffer); 3648} 3649 3650// AudioBufferProvider interface 3651status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3652 AudioBufferProvider::Buffer* buffer, int64_t pts) 3653{ 3654 audio_track_cblk_t* cblk = this->cblk(); 3655 uint32_t framesReady; 3656 uint32_t framesReq = buffer->frameCount; 3657 3658 // Check if last stepServer failed, try to step now 3659 if (mStepServerFailed) { 3660 if (!step()) goto getNextBuffer_exit; 3661 ALOGV("stepServer recovered"); 3662 mStepServerFailed = false; 3663 } 3664 3665 framesReady = cblk->framesReady(); 3666 3667 if (CC_LIKELY(framesReady)) { 3668 uint32_t s = cblk->server; 3669 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3670 3671 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3672 if (framesReq > framesReady) { 3673 framesReq = framesReady; 3674 } 3675 if (s + framesReq > bufferEnd) { 3676 framesReq = bufferEnd - s; 3677 } 3678 3679 buffer->raw = getBuffer(s, framesReq); 3680 if (buffer->raw == NULL) goto getNextBuffer_exit; 3681 3682 buffer->frameCount = framesReq; 3683 return NO_ERROR; 3684 } 3685 3686getNextBuffer_exit: 3687 buffer->raw = NULL; 3688 buffer->frameCount = 0; 3689 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3690 return NOT_ENOUGH_DATA; 3691} 3692 3693uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3694 return mCblk->framesReady(); 3695} 3696 3697bool AudioFlinger::PlaybackThread::Track::isReady() const { 3698 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3699 3700 if (framesReady() >= mCblk->frameCount || 3701 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3702 mFillingUpStatus = FS_FILLED; 3703 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3704 return true; 3705 } 3706 return false; 3707} 3708 3709status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3710{ 3711 status_t status = NO_ERROR; 3712 ALOGV("start(%d), calling pid %d session %d tid %d", 3713 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3714 sp<ThreadBase> thread = mThread.promote(); 3715 if (thread != 0) { 3716 Mutex::Autolock _l(thread->mLock); 3717 track_state state = mState; 3718 // here the track could be either new, or restarted 3719 // in both cases "unstop" the track 3720 if (mState == PAUSED) { 3721 mState = TrackBase::RESUMING; 3722 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3723 } else { 3724 mState = TrackBase::ACTIVE; 3725 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3726 } 3727 3728 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3729 thread->mLock.unlock(); 3730 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3731 thread->mLock.lock(); 3732 3733 // to track the speaker usage 3734 if (status == NO_ERROR) { 3735 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3736 } 3737 } 3738 if (status == NO_ERROR) { 3739 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3740 playbackThread->addTrack_l(this); 3741 } else { 3742 mState = state; 3743 } 3744 } else { 3745 status = BAD_VALUE; 3746 } 3747 return status; 3748} 3749 3750void AudioFlinger::PlaybackThread::Track::stop() 3751{ 3752 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3753 sp<ThreadBase> thread = mThread.promote(); 3754 if (thread != 0) { 3755 Mutex::Autolock _l(thread->mLock); 3756 track_state state = mState; 3757 if (mState > STOPPED) { 3758 mState = STOPPED; 3759 // If the track is not active (PAUSED and buffers full), flush buffers 3760 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3761 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3762 reset(); 3763 } 3764 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3765 } 3766 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3767 thread->mLock.unlock(); 3768 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3769 thread->mLock.lock(); 3770 3771 // to track the speaker usage 3772 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3773 } 3774 } 3775} 3776 3777void AudioFlinger::PlaybackThread::Track::pause() 3778{ 3779 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3780 sp<ThreadBase> thread = mThread.promote(); 3781 if (thread != 0) { 3782 Mutex::Autolock _l(thread->mLock); 3783 if (mState == ACTIVE || mState == RESUMING) { 3784 mState = PAUSING; 3785 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3786 if (!isOutputTrack()) { 3787 thread->mLock.unlock(); 3788 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3789 thread->mLock.lock(); 3790 3791 // to track the speaker usage 3792 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3793 } 3794 } 3795 } 3796} 3797 3798void AudioFlinger::PlaybackThread::Track::flush() 3799{ 3800 ALOGV("flush(%d)", mName); 3801 sp<ThreadBase> thread = mThread.promote(); 3802 if (thread != 0) { 3803 Mutex::Autolock _l(thread->mLock); 3804 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3805 return; 3806 } 3807 // No point remaining in PAUSED state after a flush => go to 3808 // STOPPED state 3809 mState = STOPPED; 3810 3811 // do not reset the track if it is still in the process of being stopped or paused. 3812 // this will be done by prepareTracks_l() when the track is stopped. 3813 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3814 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3815 reset(); 3816 } 3817 } 3818} 3819 3820void AudioFlinger::PlaybackThread::Track::reset() 3821{ 3822 // Do not reset twice to avoid discarding data written just after a flush and before 3823 // the audioflinger thread detects the track is stopped. 3824 if (!mResetDone) { 3825 TrackBase::reset(); 3826 // Force underrun condition to avoid false underrun callback until first data is 3827 // written to buffer 3828 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3829 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3830 mFillingUpStatus = FS_FILLING; 3831 mResetDone = true; 3832 } 3833} 3834 3835void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3836{ 3837 mMute = muted; 3838} 3839 3840status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3841{ 3842 status_t status = DEAD_OBJECT; 3843 sp<ThreadBase> thread = mThread.promote(); 3844 if (thread != 0) { 3845 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3846 status = playbackThread->attachAuxEffect(this, EffectId); 3847 } 3848 return status; 3849} 3850 3851void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3852{ 3853 mAuxEffectId = EffectId; 3854 mAuxBuffer = buffer; 3855} 3856 3857// timed audio tracks 3858 3859sp<AudioFlinger::PlaybackThread::TimedTrack> 3860AudioFlinger::PlaybackThread::TimedTrack::create( 3861 PlaybackThread *thread, 3862 const sp<Client>& client, 3863 audio_stream_type_t streamType, 3864 uint32_t sampleRate, 3865 audio_format_t format, 3866 uint32_t channelMask, 3867 int frameCount, 3868 const sp<IMemory>& sharedBuffer, 3869 int sessionId) { 3870 if (!client->reserveTimedTrack()) 3871 return NULL; 3872 3873 sp<TimedTrack> track = new TimedTrack( 3874 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3875 sharedBuffer, sessionId); 3876 3877 if (track == NULL) { 3878 client->releaseTimedTrack(); 3879 return NULL; 3880 } 3881 3882 return track; 3883} 3884 3885AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3886 PlaybackThread *thread, 3887 const sp<Client>& client, 3888 audio_stream_type_t streamType, 3889 uint32_t sampleRate, 3890 audio_format_t format, 3891 uint32_t channelMask, 3892 int frameCount, 3893 const sp<IMemory>& sharedBuffer, 3894 int sessionId) 3895 : Track(thread, client, streamType, sampleRate, format, channelMask, 3896 frameCount, sharedBuffer, sessionId), 3897 mTimedSilenceBuffer(NULL), 3898 mTimedSilenceBufferSize(0), 3899 mTimedAudioOutputOnTime(false), 3900 mMediaTimeTransformValid(false) 3901{ 3902 LocalClock lc; 3903 mLocalTimeFreq = lc.getLocalFreq(); 3904 3905 mLocalTimeToSampleTransform.a_zero = 0; 3906 mLocalTimeToSampleTransform.b_zero = 0; 3907 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3908 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3909 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3910 &mLocalTimeToSampleTransform.a_to_b_denom); 3911} 3912 3913AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3914 mClient->releaseTimedTrack(); 3915 delete [] mTimedSilenceBuffer; 3916} 3917 3918status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3919 size_t size, sp<IMemory>* buffer) { 3920 3921 Mutex::Autolock _l(mTimedBufferQueueLock); 3922 3923 trimTimedBufferQueue_l(); 3924 3925 // lazily initialize the shared memory heap for timed buffers 3926 if (mTimedMemoryDealer == NULL) { 3927 const int kTimedBufferHeapSize = 512 << 10; 3928 3929 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3930 "AudioFlingerTimed"); 3931 if (mTimedMemoryDealer == NULL) 3932 return NO_MEMORY; 3933 } 3934 3935 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3936 if (newBuffer == NULL) { 3937 newBuffer = mTimedMemoryDealer->allocate(size); 3938 if (newBuffer == NULL) 3939 return NO_MEMORY; 3940 } 3941 3942 *buffer = newBuffer; 3943 return NO_ERROR; 3944} 3945 3946// caller must hold mTimedBufferQueueLock 3947void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3948 int64_t mediaTimeNow; 3949 { 3950 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3951 if (!mMediaTimeTransformValid) 3952 return; 3953 3954 int64_t targetTimeNow; 3955 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3956 ? mCCHelper.getCommonTime(&targetTimeNow) 3957 : mCCHelper.getLocalTime(&targetTimeNow); 3958 3959 if (OK != res) 3960 return; 3961 3962 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3963 &mediaTimeNow)) { 3964 return; 3965 } 3966 } 3967 3968 size_t trimIndex; 3969 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3970 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3971 break; 3972 } 3973 3974 if (trimIndex) { 3975 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3976 } 3977} 3978 3979status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3980 const sp<IMemory>& buffer, int64_t pts) { 3981 3982 { 3983 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3984 if (!mMediaTimeTransformValid) 3985 return INVALID_OPERATION; 3986 } 3987 3988 Mutex::Autolock _l(mTimedBufferQueueLock); 3989 3990 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3991 3992 return NO_ERROR; 3993} 3994 3995status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3996 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3997 3998 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3999 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4000 target); 4001 4002 if (!(target == TimedAudioTrack::LOCAL_TIME || 4003 target == TimedAudioTrack::COMMON_TIME)) { 4004 return BAD_VALUE; 4005 } 4006 4007 Mutex::Autolock lock(mMediaTimeTransformLock); 4008 mMediaTimeTransform = xform; 4009 mMediaTimeTransformTarget = target; 4010 mMediaTimeTransformValid = true; 4011 4012 return NO_ERROR; 4013} 4014 4015#define min(a, b) ((a) < (b) ? (a) : (b)) 4016 4017// implementation of getNextBuffer for tracks whose buffers have timestamps 4018status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4019 AudioBufferProvider::Buffer* buffer, int64_t pts) 4020{ 4021 if (pts == AudioBufferProvider::kInvalidPTS) { 4022 buffer->raw = 0; 4023 buffer->frameCount = 0; 4024 return INVALID_OPERATION; 4025 } 4026 4027 Mutex::Autolock _l(mTimedBufferQueueLock); 4028 4029 while (true) { 4030 4031 // if we have no timed buffers, then fail 4032 if (mTimedBufferQueue.isEmpty()) { 4033 buffer->raw = 0; 4034 buffer->frameCount = 0; 4035 return NOT_ENOUGH_DATA; 4036 } 4037 4038 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4039 4040 // calculate the PTS of the head of the timed buffer queue expressed in 4041 // local time 4042 int64_t headLocalPTS; 4043 { 4044 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4045 4046 assert(mMediaTimeTransformValid); 4047 4048 if (mMediaTimeTransform.a_to_b_denom == 0) { 4049 // the transform represents a pause, so yield silence 4050 timedYieldSilence(buffer->frameCount, buffer); 4051 return NO_ERROR; 4052 } 4053 4054 int64_t transformedPTS; 4055 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4056 &transformedPTS)) { 4057 // the transform failed. this shouldn't happen, but if it does 4058 // then just drop this buffer 4059 ALOGW("timedGetNextBuffer transform failed"); 4060 buffer->raw = 0; 4061 buffer->frameCount = 0; 4062 mTimedBufferQueue.removeAt(0); 4063 return NO_ERROR; 4064 } 4065 4066 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4067 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4068 &headLocalPTS)) { 4069 buffer->raw = 0; 4070 buffer->frameCount = 0; 4071 return INVALID_OPERATION; 4072 } 4073 } else { 4074 headLocalPTS = transformedPTS; 4075 } 4076 } 4077 4078 // adjust the head buffer's PTS to reflect the portion of the head buffer 4079 // that has already been consumed 4080 int64_t effectivePTS = headLocalPTS + 4081 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4082 4083 // Calculate the delta in samples between the head of the input buffer 4084 // queue and the start of the next output buffer that will be written. 4085 // If the transformation fails because of over or underflow, it means 4086 // that the sample's position in the output stream is so far out of 4087 // whack that it should just be dropped. 4088 int64_t sampleDelta; 4089 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4090 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4091 mTimedBufferQueue.removeAt(0); 4092 continue; 4093 } 4094 if (!mLocalTimeToSampleTransform.doForwardTransform( 4095 (effectivePTS - pts) << 32, &sampleDelta)) { 4096 ALOGV("*** too late during sample rate transform: dropped buffer"); 4097 mTimedBufferQueue.removeAt(0); 4098 continue; 4099 } 4100 4101 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4102 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4103 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4104 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4105 4106 // if the delta between the ideal placement for the next input sample and 4107 // the current output position is within this threshold, then we will 4108 // concatenate the next input samples to the previous output 4109 const int64_t kSampleContinuityThreshold = 4110 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4111 4112 // if this is the first buffer of audio that we're emitting from this track 4113 // then it should be almost exactly on time. 4114 const int64_t kSampleStartupThreshold = 1LL << 32; 4115 4116 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4117 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4118 // the next input is close enough to being on time, so concatenate it 4119 // with the last output 4120 timedYieldSamples(buffer); 4121 4122 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4123 return NO_ERROR; 4124 } else if (sampleDelta > 0) { 4125 // the gap between the current output position and the proper start of 4126 // the next input sample is too big, so fill it with silence 4127 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4128 4129 timedYieldSilence(framesUntilNextInput, buffer); 4130 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4131 return NO_ERROR; 4132 } else { 4133 // the next input sample is late 4134 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4135 size_t onTimeSamplePosition = 4136 head.position() + lateFrames * mCblk->frameSize; 4137 4138 if (onTimeSamplePosition > head.buffer()->size()) { 4139 // all the remaining samples in the head are too late, so 4140 // drop it and move on 4141 ALOGV("*** too late: dropped buffer"); 4142 mTimedBufferQueue.removeAt(0); 4143 continue; 4144 } else { 4145 // skip over the late samples 4146 head.setPosition(onTimeSamplePosition); 4147 4148 // yield the available samples 4149 timedYieldSamples(buffer); 4150 4151 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4152 return NO_ERROR; 4153 } 4154 } 4155 } 4156} 4157 4158// Yield samples from the timed buffer queue head up to the given output 4159// buffer's capacity. 4160// 4161// Caller must hold mTimedBufferQueueLock 4162void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4163 AudioBufferProvider::Buffer* buffer) { 4164 4165 const TimedBuffer& head = mTimedBufferQueue[0]; 4166 4167 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4168 head.position()); 4169 4170 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4171 mCblk->frameSize); 4172 size_t framesRequested = buffer->frameCount; 4173 buffer->frameCount = min(framesLeftInHead, framesRequested); 4174 4175 mTimedAudioOutputOnTime = true; 4176} 4177 4178// Yield samples of silence up to the given output buffer's capacity 4179// 4180// Caller must hold mTimedBufferQueueLock 4181void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4182 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4183 4184 // lazily allocate a buffer filled with silence 4185 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4186 delete [] mTimedSilenceBuffer; 4187 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4188 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4189 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4190 } 4191 4192 buffer->raw = mTimedSilenceBuffer; 4193 size_t framesRequested = buffer->frameCount; 4194 buffer->frameCount = min(numFrames, framesRequested); 4195 4196 mTimedAudioOutputOnTime = false; 4197} 4198 4199// AudioBufferProvider interface 4200void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4201 AudioBufferProvider::Buffer* buffer) { 4202 4203 Mutex::Autolock _l(mTimedBufferQueueLock); 4204 4205 // If the buffer which was just released is part of the buffer at the head 4206 // of the queue, be sure to update the amt of the buffer which has been 4207 // consumed. If the buffer being returned is not part of the head of the 4208 // queue, its either because the buffer is part of the silence buffer, or 4209 // because the head of the timed queue was trimmed after the mixer called 4210 // getNextBuffer but before the mixer called releaseBuffer. 4211 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4212 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4213 4214 void* start = head.buffer()->pointer(); 4215 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4216 4217 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4218 head.setPosition(head.position() + 4219 (buffer->frameCount * mCblk->frameSize)); 4220 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4221 mTimedBufferQueue.removeAt(0); 4222 } 4223 } 4224 } 4225 4226 buffer->raw = 0; 4227 buffer->frameCount = 0; 4228} 4229 4230uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4231 Mutex::Autolock _l(mTimedBufferQueueLock); 4232 4233 uint32_t frames = 0; 4234 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4235 const TimedBuffer& tb = mTimedBufferQueue[i]; 4236 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4237 } 4238 4239 return frames; 4240} 4241 4242AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4243 : mPTS(0), mPosition(0) {} 4244 4245AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4246 const sp<IMemory>& buffer, int64_t pts) 4247 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4248 4249// ---------------------------------------------------------------------------- 4250 4251// RecordTrack constructor must be called with AudioFlinger::mLock held 4252AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4253 RecordThread *thread, 4254 const sp<Client>& client, 4255 uint32_t sampleRate, 4256 audio_format_t format, 4257 uint32_t channelMask, 4258 int frameCount, 4259 int sessionId) 4260 : TrackBase(thread, client, sampleRate, format, 4261 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4262 mOverflow(false) 4263{ 4264 if (mCblk != NULL) { 4265 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4266 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4267 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4268 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4269 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4270 } else { 4271 mCblk->frameSize = sizeof(int8_t); 4272 } 4273 } 4274} 4275 4276AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4277{ 4278 sp<ThreadBase> thread = mThread.promote(); 4279 if (thread != 0) { 4280 AudioSystem::releaseInput(thread->id()); 4281 } 4282} 4283 4284// AudioBufferProvider interface 4285status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4286{ 4287 audio_track_cblk_t* cblk = this->cblk(); 4288 uint32_t framesAvail; 4289 uint32_t framesReq = buffer->frameCount; 4290 4291 // Check if last stepServer failed, try to step now 4292 if (mStepServerFailed) { 4293 if (!step()) goto getNextBuffer_exit; 4294 ALOGV("stepServer recovered"); 4295 mStepServerFailed = false; 4296 } 4297 4298 framesAvail = cblk->framesAvailable_l(); 4299 4300 if (CC_LIKELY(framesAvail)) { 4301 uint32_t s = cblk->server; 4302 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4303 4304 if (framesReq > framesAvail) { 4305 framesReq = framesAvail; 4306 } 4307 if (s + framesReq > bufferEnd) { 4308 framesReq = bufferEnd - s; 4309 } 4310 4311 buffer->raw = getBuffer(s, framesReq); 4312 if (buffer->raw == NULL) goto getNextBuffer_exit; 4313 4314 buffer->frameCount = framesReq; 4315 return NO_ERROR; 4316 } 4317 4318getNextBuffer_exit: 4319 buffer->raw = NULL; 4320 buffer->frameCount = 0; 4321 return NOT_ENOUGH_DATA; 4322} 4323 4324status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4325{ 4326 sp<ThreadBase> thread = mThread.promote(); 4327 if (thread != 0) { 4328 RecordThread *recordThread = (RecordThread *)thread.get(); 4329 return recordThread->start(this, tid); 4330 } else { 4331 return BAD_VALUE; 4332 } 4333} 4334 4335void AudioFlinger::RecordThread::RecordTrack::stop() 4336{ 4337 sp<ThreadBase> thread = mThread.promote(); 4338 if (thread != 0) { 4339 RecordThread *recordThread = (RecordThread *)thread.get(); 4340 recordThread->stop(this); 4341 TrackBase::reset(); 4342 // Force overerrun condition to avoid false overrun callback until first data is 4343 // read from buffer 4344 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4345 } 4346} 4347 4348void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4349{ 4350 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4351 (mClient == 0) ? getpid_cached : mClient->pid(), 4352 mFormat, 4353 mChannelMask, 4354 mSessionId, 4355 mFrameCount, 4356 mState, 4357 mCblk->sampleRate, 4358 mCblk->server, 4359 mCblk->user); 4360} 4361 4362 4363// ---------------------------------------------------------------------------- 4364 4365AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4366 PlaybackThread *playbackThread, 4367 DuplicatingThread *sourceThread, 4368 uint32_t sampleRate, 4369 audio_format_t format, 4370 uint32_t channelMask, 4371 int frameCount) 4372 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4373 mActive(false), mSourceThread(sourceThread) 4374{ 4375 4376 if (mCblk != NULL) { 4377 mCblk->flags |= CBLK_DIRECTION_OUT; 4378 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4379 mOutBuffer.frameCount = 0; 4380 playbackThread->mTracks.add(this); 4381 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4382 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4383 mCblk, mBuffer, mCblk->buffers, 4384 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4385 } else { 4386 ALOGW("Error creating output track on thread %p", playbackThread); 4387 } 4388} 4389 4390AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4391{ 4392 clearBufferQueue(); 4393} 4394 4395status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4396{ 4397 status_t status = Track::start(tid); 4398 if (status != NO_ERROR) { 4399 return status; 4400 } 4401 4402 mActive = true; 4403 mRetryCount = 127; 4404 return status; 4405} 4406 4407void AudioFlinger::PlaybackThread::OutputTrack::stop() 4408{ 4409 Track::stop(); 4410 clearBufferQueue(); 4411 mOutBuffer.frameCount = 0; 4412 mActive = false; 4413} 4414 4415bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4416{ 4417 Buffer *pInBuffer; 4418 Buffer inBuffer; 4419 uint32_t channelCount = mChannelCount; 4420 bool outputBufferFull = false; 4421 inBuffer.frameCount = frames; 4422 inBuffer.i16 = data; 4423 4424 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4425 4426 if (!mActive && frames != 0) { 4427 start(0); 4428 sp<ThreadBase> thread = mThread.promote(); 4429 if (thread != 0) { 4430 MixerThread *mixerThread = (MixerThread *)thread.get(); 4431 if (mCblk->frameCount > frames){ 4432 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4433 uint32_t startFrames = (mCblk->frameCount - frames); 4434 pInBuffer = new Buffer; 4435 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4436 pInBuffer->frameCount = startFrames; 4437 pInBuffer->i16 = pInBuffer->mBuffer; 4438 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4439 mBufferQueue.add(pInBuffer); 4440 } else { 4441 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4442 } 4443 } 4444 } 4445 } 4446 4447 while (waitTimeLeftMs) { 4448 // First write pending buffers, then new data 4449 if (mBufferQueue.size()) { 4450 pInBuffer = mBufferQueue.itemAt(0); 4451 } else { 4452 pInBuffer = &inBuffer; 4453 } 4454 4455 if (pInBuffer->frameCount == 0) { 4456 break; 4457 } 4458 4459 if (mOutBuffer.frameCount == 0) { 4460 mOutBuffer.frameCount = pInBuffer->frameCount; 4461 nsecs_t startTime = systemTime(); 4462 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4463 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4464 outputBufferFull = true; 4465 break; 4466 } 4467 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4468 if (waitTimeLeftMs >= waitTimeMs) { 4469 waitTimeLeftMs -= waitTimeMs; 4470 } else { 4471 waitTimeLeftMs = 0; 4472 } 4473 } 4474 4475 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4476 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4477 mCblk->stepUser(outFrames); 4478 pInBuffer->frameCount -= outFrames; 4479 pInBuffer->i16 += outFrames * channelCount; 4480 mOutBuffer.frameCount -= outFrames; 4481 mOutBuffer.i16 += outFrames * channelCount; 4482 4483 if (pInBuffer->frameCount == 0) { 4484 if (mBufferQueue.size()) { 4485 mBufferQueue.removeAt(0); 4486 delete [] pInBuffer->mBuffer; 4487 delete pInBuffer; 4488 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4489 } else { 4490 break; 4491 } 4492 } 4493 } 4494 4495 // If we could not write all frames, allocate a buffer and queue it for next time. 4496 if (inBuffer.frameCount) { 4497 sp<ThreadBase> thread = mThread.promote(); 4498 if (thread != 0 && !thread->standby()) { 4499 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4500 pInBuffer = new Buffer; 4501 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4502 pInBuffer->frameCount = inBuffer.frameCount; 4503 pInBuffer->i16 = pInBuffer->mBuffer; 4504 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4505 mBufferQueue.add(pInBuffer); 4506 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4507 } else { 4508 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4509 } 4510 } 4511 } 4512 4513 // Calling write() with a 0 length buffer, means that no more data will be written: 4514 // If no more buffers are pending, fill output track buffer to make sure it is started 4515 // by output mixer. 4516 if (frames == 0 && mBufferQueue.size() == 0) { 4517 if (mCblk->user < mCblk->frameCount) { 4518 frames = mCblk->frameCount - mCblk->user; 4519 pInBuffer = new Buffer; 4520 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4521 pInBuffer->frameCount = frames; 4522 pInBuffer->i16 = pInBuffer->mBuffer; 4523 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4524 mBufferQueue.add(pInBuffer); 4525 } else if (mActive) { 4526 stop(); 4527 } 4528 } 4529 4530 return outputBufferFull; 4531} 4532 4533status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4534{ 4535 int active; 4536 status_t result; 4537 audio_track_cblk_t* cblk = mCblk; 4538 uint32_t framesReq = buffer->frameCount; 4539 4540// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4541 buffer->frameCount = 0; 4542 4543 uint32_t framesAvail = cblk->framesAvailable(); 4544 4545 4546 if (framesAvail == 0) { 4547 Mutex::Autolock _l(cblk->lock); 4548 goto start_loop_here; 4549 while (framesAvail == 0) { 4550 active = mActive; 4551 if (CC_UNLIKELY(!active)) { 4552 ALOGV("Not active and NO_MORE_BUFFERS"); 4553 return NO_MORE_BUFFERS; 4554 } 4555 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4556 if (result != NO_ERROR) { 4557 return NO_MORE_BUFFERS; 4558 } 4559 // read the server count again 4560 start_loop_here: 4561 framesAvail = cblk->framesAvailable_l(); 4562 } 4563 } 4564 4565// if (framesAvail < framesReq) { 4566// return NO_MORE_BUFFERS; 4567// } 4568 4569 if (framesReq > framesAvail) { 4570 framesReq = framesAvail; 4571 } 4572 4573 uint32_t u = cblk->user; 4574 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4575 4576 if (u + framesReq > bufferEnd) { 4577 framesReq = bufferEnd - u; 4578 } 4579 4580 buffer->frameCount = framesReq; 4581 buffer->raw = (void *)cblk->buffer(u); 4582 return NO_ERROR; 4583} 4584 4585 4586void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4587{ 4588 size_t size = mBufferQueue.size(); 4589 4590 for (size_t i = 0; i < size; i++) { 4591 Buffer *pBuffer = mBufferQueue.itemAt(i); 4592 delete [] pBuffer->mBuffer; 4593 delete pBuffer; 4594 } 4595 mBufferQueue.clear(); 4596} 4597 4598// ---------------------------------------------------------------------------- 4599 4600AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4601 : RefBase(), 4602 mAudioFlinger(audioFlinger), 4603 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4604 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4605 mPid(pid), 4606 mTimedTrackCount(0) 4607{ 4608 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4609} 4610 4611// Client destructor must be called with AudioFlinger::mLock held 4612AudioFlinger::Client::~Client() 4613{ 4614 mAudioFlinger->removeClient_l(mPid); 4615} 4616 4617sp<MemoryDealer> AudioFlinger::Client::heap() const 4618{ 4619 return mMemoryDealer; 4620} 4621 4622// Reserve one of the limited slots for a timed audio track associated 4623// with this client 4624bool AudioFlinger::Client::reserveTimedTrack() 4625{ 4626 const int kMaxTimedTracksPerClient = 4; 4627 4628 Mutex::Autolock _l(mTimedTrackLock); 4629 4630 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4631 ALOGW("can not create timed track - pid %d has exceeded the limit", 4632 mPid); 4633 return false; 4634 } 4635 4636 mTimedTrackCount++; 4637 return true; 4638} 4639 4640// Release a slot for a timed audio track 4641void AudioFlinger::Client::releaseTimedTrack() 4642{ 4643 Mutex::Autolock _l(mTimedTrackLock); 4644 mTimedTrackCount--; 4645} 4646 4647// ---------------------------------------------------------------------------- 4648 4649AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4650 const sp<IAudioFlingerClient>& client, 4651 pid_t pid) 4652 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4653{ 4654} 4655 4656AudioFlinger::NotificationClient::~NotificationClient() 4657{ 4658} 4659 4660void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4661{ 4662 sp<NotificationClient> keep(this); 4663 mAudioFlinger->removeNotificationClient(mPid); 4664} 4665 4666// ---------------------------------------------------------------------------- 4667 4668AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4669 : BnAudioTrack(), 4670 mTrack(track) 4671{ 4672} 4673 4674AudioFlinger::TrackHandle::~TrackHandle() { 4675 // just stop the track on deletion, associated resources 4676 // will be freed from the main thread once all pending buffers have 4677 // been played. Unless it's not in the active track list, in which 4678 // case we free everything now... 4679 mTrack->destroy(); 4680} 4681 4682sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4683 return mTrack->getCblk(); 4684} 4685 4686status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4687 return mTrack->start(tid); 4688} 4689 4690void AudioFlinger::TrackHandle::stop() { 4691 mTrack->stop(); 4692} 4693 4694void AudioFlinger::TrackHandle::flush() { 4695 mTrack->flush(); 4696} 4697 4698void AudioFlinger::TrackHandle::mute(bool e) { 4699 mTrack->mute(e); 4700} 4701 4702void AudioFlinger::TrackHandle::pause() { 4703 mTrack->pause(); 4704} 4705 4706status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4707{ 4708 return mTrack->attachAuxEffect(EffectId); 4709} 4710 4711status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4712 sp<IMemory>* buffer) { 4713 if (!mTrack->isTimedTrack()) 4714 return INVALID_OPERATION; 4715 4716 PlaybackThread::TimedTrack* tt = 4717 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4718 return tt->allocateTimedBuffer(size, buffer); 4719} 4720 4721status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4722 int64_t pts) { 4723 if (!mTrack->isTimedTrack()) 4724 return INVALID_OPERATION; 4725 4726 PlaybackThread::TimedTrack* tt = 4727 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4728 return tt->queueTimedBuffer(buffer, pts); 4729} 4730 4731status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4732 const LinearTransform& xform, int target) { 4733 4734 if (!mTrack->isTimedTrack()) 4735 return INVALID_OPERATION; 4736 4737 PlaybackThread::TimedTrack* tt = 4738 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4739 return tt->setMediaTimeTransform( 4740 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4741} 4742 4743status_t AudioFlinger::TrackHandle::onTransact( 4744 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4745{ 4746 return BnAudioTrack::onTransact(code, data, reply, flags); 4747} 4748 4749// ---------------------------------------------------------------------------- 4750 4751sp<IAudioRecord> AudioFlinger::openRecord( 4752 pid_t pid, 4753 audio_io_handle_t input, 4754 uint32_t sampleRate, 4755 audio_format_t format, 4756 uint32_t channelMask, 4757 int frameCount, 4758 // FIXME dead, remove from IAudioFlinger 4759 uint32_t flags, 4760 int *sessionId, 4761 status_t *status) 4762{ 4763 sp<RecordThread::RecordTrack> recordTrack; 4764 sp<RecordHandle> recordHandle; 4765 sp<Client> client; 4766 status_t lStatus; 4767 RecordThread *thread; 4768 size_t inFrameCount; 4769 int lSessionId; 4770 4771 // check calling permissions 4772 if (!recordingAllowed()) { 4773 lStatus = PERMISSION_DENIED; 4774 goto Exit; 4775 } 4776 4777 // add client to list 4778 { // scope for mLock 4779 Mutex::Autolock _l(mLock); 4780 thread = checkRecordThread_l(input); 4781 if (thread == NULL) { 4782 lStatus = BAD_VALUE; 4783 goto Exit; 4784 } 4785 4786 client = registerPid_l(pid); 4787 4788 // If no audio session id is provided, create one here 4789 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4790 lSessionId = *sessionId; 4791 } else { 4792 lSessionId = nextUniqueId(); 4793 if (sessionId != NULL) { 4794 *sessionId = lSessionId; 4795 } 4796 } 4797 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4798 recordTrack = thread->createRecordTrack_l(client, 4799 sampleRate, 4800 format, 4801 channelMask, 4802 frameCount, 4803 lSessionId, 4804 &lStatus); 4805 } 4806 if (lStatus != NO_ERROR) { 4807 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4808 // destructor is called by the TrackBase destructor with mLock held 4809 client.clear(); 4810 recordTrack.clear(); 4811 goto Exit; 4812 } 4813 4814 // return to handle to client 4815 recordHandle = new RecordHandle(recordTrack); 4816 lStatus = NO_ERROR; 4817 4818Exit: 4819 if (status) { 4820 *status = lStatus; 4821 } 4822 return recordHandle; 4823} 4824 4825// ---------------------------------------------------------------------------- 4826 4827AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4828 : BnAudioRecord(), 4829 mRecordTrack(recordTrack) 4830{ 4831} 4832 4833AudioFlinger::RecordHandle::~RecordHandle() { 4834 stop(); 4835} 4836 4837sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4838 return mRecordTrack->getCblk(); 4839} 4840 4841status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4842 ALOGV("RecordHandle::start()"); 4843 return mRecordTrack->start(tid); 4844} 4845 4846void AudioFlinger::RecordHandle::stop() { 4847 ALOGV("RecordHandle::stop()"); 4848 mRecordTrack->stop(); 4849} 4850 4851status_t AudioFlinger::RecordHandle::onTransact( 4852 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4853{ 4854 return BnAudioRecord::onTransact(code, data, reply, flags); 4855} 4856 4857// ---------------------------------------------------------------------------- 4858 4859AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4860 AudioStreamIn *input, 4861 uint32_t sampleRate, 4862 uint32_t channels, 4863 audio_io_handle_t id, 4864 uint32_t device) : 4865 ThreadBase(audioFlinger, id, device, RECORD), 4866 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4867 // mRsmpInIndex and mInputBytes set by readInputParameters() 4868 mReqChannelCount(popcount(channels)), 4869 mReqSampleRate(sampleRate) 4870 // mBytesRead is only meaningful while active, and so is cleared in start() 4871 // (but might be better to also clear here for dump?) 4872{ 4873 snprintf(mName, kNameLength, "AudioIn_%d", id); 4874 4875 readInputParameters(); 4876} 4877 4878 4879AudioFlinger::RecordThread::~RecordThread() 4880{ 4881 delete[] mRsmpInBuffer; 4882 delete mResampler; 4883 delete[] mRsmpOutBuffer; 4884} 4885 4886void AudioFlinger::RecordThread::onFirstRef() 4887{ 4888 run(mName, PRIORITY_URGENT_AUDIO); 4889} 4890 4891status_t AudioFlinger::RecordThread::readyToRun() 4892{ 4893 status_t status = initCheck(); 4894 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4895 return status; 4896} 4897 4898bool AudioFlinger::RecordThread::threadLoop() 4899{ 4900 AudioBufferProvider::Buffer buffer; 4901 sp<RecordTrack> activeTrack; 4902 Vector< sp<EffectChain> > effectChains; 4903 4904 nsecs_t lastWarning = 0; 4905 4906 acquireWakeLock(); 4907 4908 // start recording 4909 while (!exitPending()) { 4910 4911 processConfigEvents(); 4912 4913 { // scope for mLock 4914 Mutex::Autolock _l(mLock); 4915 checkForNewParameters_l(); 4916 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4917 if (!mStandby) { 4918 mInput->stream->common.standby(&mInput->stream->common); 4919 mStandby = true; 4920 } 4921 4922 if (exitPending()) break; 4923 4924 releaseWakeLock_l(); 4925 ALOGV("RecordThread: loop stopping"); 4926 // go to sleep 4927 mWaitWorkCV.wait(mLock); 4928 ALOGV("RecordThread: loop starting"); 4929 acquireWakeLock_l(); 4930 continue; 4931 } 4932 if (mActiveTrack != 0) { 4933 if (mActiveTrack->mState == TrackBase::PAUSING) { 4934 if (!mStandby) { 4935 mInput->stream->common.standby(&mInput->stream->common); 4936 mStandby = true; 4937 } 4938 mActiveTrack.clear(); 4939 mStartStopCond.broadcast(); 4940 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4941 if (mReqChannelCount != mActiveTrack->channelCount()) { 4942 mActiveTrack.clear(); 4943 mStartStopCond.broadcast(); 4944 } else if (mBytesRead != 0) { 4945 // record start succeeds only if first read from audio input 4946 // succeeds 4947 if (mBytesRead > 0) { 4948 mActiveTrack->mState = TrackBase::ACTIVE; 4949 } else { 4950 mActiveTrack.clear(); 4951 } 4952 mStartStopCond.broadcast(); 4953 } 4954 mStandby = false; 4955 } 4956 } 4957 lockEffectChains_l(effectChains); 4958 } 4959 4960 if (mActiveTrack != 0) { 4961 if (mActiveTrack->mState != TrackBase::ACTIVE && 4962 mActiveTrack->mState != TrackBase::RESUMING) { 4963 unlockEffectChains(effectChains); 4964 usleep(kRecordThreadSleepUs); 4965 continue; 4966 } 4967 for (size_t i = 0; i < effectChains.size(); i ++) { 4968 effectChains[i]->process_l(); 4969 } 4970 4971 buffer.frameCount = mFrameCount; 4972 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4973 size_t framesOut = buffer.frameCount; 4974 if (mResampler == NULL) { 4975 // no resampling 4976 while (framesOut) { 4977 size_t framesIn = mFrameCount - mRsmpInIndex; 4978 if (framesIn) { 4979 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4980 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4981 if (framesIn > framesOut) 4982 framesIn = framesOut; 4983 mRsmpInIndex += framesIn; 4984 framesOut -= framesIn; 4985 if ((int)mChannelCount == mReqChannelCount || 4986 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4987 memcpy(dst, src, framesIn * mFrameSize); 4988 } else { 4989 int16_t *src16 = (int16_t *)src; 4990 int16_t *dst16 = (int16_t *)dst; 4991 if (mChannelCount == 1) { 4992 while (framesIn--) { 4993 *dst16++ = *src16; 4994 *dst16++ = *src16++; 4995 } 4996 } else { 4997 while (framesIn--) { 4998 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4999 src16 += 2; 5000 } 5001 } 5002 } 5003 } 5004 if (framesOut && mFrameCount == mRsmpInIndex) { 5005 if (framesOut == mFrameCount && 5006 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5007 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5008 framesOut = 0; 5009 } else { 5010 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5011 mRsmpInIndex = 0; 5012 } 5013 if (mBytesRead < 0) { 5014 ALOGE("Error reading audio input"); 5015 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5016 // Force input into standby so that it tries to 5017 // recover at next read attempt 5018 mInput->stream->common.standby(&mInput->stream->common); 5019 usleep(kRecordThreadSleepUs); 5020 } 5021 mRsmpInIndex = mFrameCount; 5022 framesOut = 0; 5023 buffer.frameCount = 0; 5024 } 5025 } 5026 } 5027 } else { 5028 // resampling 5029 5030 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5031 // alter output frame count as if we were expecting stereo samples 5032 if (mChannelCount == 1 && mReqChannelCount == 1) { 5033 framesOut >>= 1; 5034 } 5035 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5036 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5037 // are 32 bit aligned which should be always true. 5038 if (mChannelCount == 2 && mReqChannelCount == 1) { 5039 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5040 // the resampler always outputs stereo samples: do post stereo to mono conversion 5041 int16_t *src = (int16_t *)mRsmpOutBuffer; 5042 int16_t *dst = buffer.i16; 5043 while (framesOut--) { 5044 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5045 src += 2; 5046 } 5047 } else { 5048 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5049 } 5050 5051 } 5052 mActiveTrack->releaseBuffer(&buffer); 5053 mActiveTrack->overflow(); 5054 } 5055 // client isn't retrieving buffers fast enough 5056 else { 5057 if (!mActiveTrack->setOverflow()) { 5058 nsecs_t now = systemTime(); 5059 if ((now - lastWarning) > kWarningThrottleNs) { 5060 ALOGW("RecordThread: buffer overflow"); 5061 lastWarning = now; 5062 } 5063 } 5064 // Release the processor for a while before asking for a new buffer. 5065 // This will give the application more chance to read from the buffer and 5066 // clear the overflow. 5067 usleep(kRecordThreadSleepUs); 5068 } 5069 } 5070 // enable changes in effect chain 5071 unlockEffectChains(effectChains); 5072 effectChains.clear(); 5073 } 5074 5075 if (!mStandby) { 5076 mInput->stream->common.standby(&mInput->stream->common); 5077 } 5078 mActiveTrack.clear(); 5079 5080 mStartStopCond.broadcast(); 5081 5082 releaseWakeLock(); 5083 5084 ALOGV("RecordThread %p exiting", this); 5085 return false; 5086} 5087 5088 5089sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5090 const sp<AudioFlinger::Client>& client, 5091 uint32_t sampleRate, 5092 audio_format_t format, 5093 int channelMask, 5094 int frameCount, 5095 int sessionId, 5096 status_t *status) 5097{ 5098 sp<RecordTrack> track; 5099 status_t lStatus; 5100 5101 lStatus = initCheck(); 5102 if (lStatus != NO_ERROR) { 5103 ALOGE("Audio driver not initialized."); 5104 goto Exit; 5105 } 5106 5107 { // scope for mLock 5108 Mutex::Autolock _l(mLock); 5109 5110 track = new RecordTrack(this, client, sampleRate, 5111 format, channelMask, frameCount, sessionId); 5112 5113 if (track->getCblk() == 0) { 5114 lStatus = NO_MEMORY; 5115 goto Exit; 5116 } 5117 5118 mTrack = track.get(); 5119 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5120 bool suspend = audio_is_bluetooth_sco_device( 5121 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5122 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5123 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5124 } 5125 lStatus = NO_ERROR; 5126 5127Exit: 5128 if (status) { 5129 *status = lStatus; 5130 } 5131 return track; 5132} 5133 5134status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5135{ 5136 ALOGV("RecordThread::start tid=%d", tid); 5137 sp <ThreadBase> strongMe = this; 5138 status_t status = NO_ERROR; 5139 { 5140 AutoMutex lock(mLock); 5141 if (mActiveTrack != 0) { 5142 if (recordTrack != mActiveTrack.get()) { 5143 status = -EBUSY; 5144 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5145 mActiveTrack->mState = TrackBase::ACTIVE; 5146 } 5147 return status; 5148 } 5149 5150 recordTrack->mState = TrackBase::IDLE; 5151 mActiveTrack = recordTrack; 5152 mLock.unlock(); 5153 status_t status = AudioSystem::startInput(mId); 5154 mLock.lock(); 5155 if (status != NO_ERROR) { 5156 mActiveTrack.clear(); 5157 return status; 5158 } 5159 mRsmpInIndex = mFrameCount; 5160 mBytesRead = 0; 5161 if (mResampler != NULL) { 5162 mResampler->reset(); 5163 } 5164 mActiveTrack->mState = TrackBase::RESUMING; 5165 // signal thread to start 5166 ALOGV("Signal record thread"); 5167 mWaitWorkCV.signal(); 5168 // do not wait for mStartStopCond if exiting 5169 if (exitPending()) { 5170 mActiveTrack.clear(); 5171 status = INVALID_OPERATION; 5172 goto startError; 5173 } 5174 mStartStopCond.wait(mLock); 5175 if (mActiveTrack == 0) { 5176 ALOGV("Record failed to start"); 5177 status = BAD_VALUE; 5178 goto startError; 5179 } 5180 ALOGV("Record started OK"); 5181 return status; 5182 } 5183startError: 5184 AudioSystem::stopInput(mId); 5185 return status; 5186} 5187 5188void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5189 ALOGV("RecordThread::stop"); 5190 sp <ThreadBase> strongMe = this; 5191 { 5192 AutoMutex lock(mLock); 5193 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5194 mActiveTrack->mState = TrackBase::PAUSING; 5195 // do not wait for mStartStopCond if exiting 5196 if (exitPending()) { 5197 return; 5198 } 5199 mStartStopCond.wait(mLock); 5200 // if we have been restarted, recordTrack == mActiveTrack.get() here 5201 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5202 mLock.unlock(); 5203 AudioSystem::stopInput(mId); 5204 mLock.lock(); 5205 ALOGV("Record stopped OK"); 5206 } 5207 } 5208 } 5209} 5210 5211status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5212{ 5213 const size_t SIZE = 256; 5214 char buffer[SIZE]; 5215 String8 result; 5216 5217 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5218 result.append(buffer); 5219 5220 if (mActiveTrack != 0) { 5221 result.append("Active Track:\n"); 5222 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5223 mActiveTrack->dump(buffer, SIZE); 5224 result.append(buffer); 5225 5226 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5227 result.append(buffer); 5228 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5229 result.append(buffer); 5230 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5231 result.append(buffer); 5232 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5233 result.append(buffer); 5234 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5235 result.append(buffer); 5236 5237 5238 } else { 5239 result.append("No record client\n"); 5240 } 5241 write(fd, result.string(), result.size()); 5242 5243 dumpBase(fd, args); 5244 dumpEffectChains(fd, args); 5245 5246 return NO_ERROR; 5247} 5248 5249// AudioBufferProvider interface 5250status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5251{ 5252 size_t framesReq = buffer->frameCount; 5253 size_t framesReady = mFrameCount - mRsmpInIndex; 5254 int channelCount; 5255 5256 if (framesReady == 0) { 5257 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5258 if (mBytesRead < 0) { 5259 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5260 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5261 // Force input into standby so that it tries to 5262 // recover at next read attempt 5263 mInput->stream->common.standby(&mInput->stream->common); 5264 usleep(kRecordThreadSleepUs); 5265 } 5266 buffer->raw = NULL; 5267 buffer->frameCount = 0; 5268 return NOT_ENOUGH_DATA; 5269 } 5270 mRsmpInIndex = 0; 5271 framesReady = mFrameCount; 5272 } 5273 5274 if (framesReq > framesReady) { 5275 framesReq = framesReady; 5276 } 5277 5278 if (mChannelCount == 1 && mReqChannelCount == 2) { 5279 channelCount = 1; 5280 } else { 5281 channelCount = 2; 5282 } 5283 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5284 buffer->frameCount = framesReq; 5285 return NO_ERROR; 5286} 5287 5288// AudioBufferProvider interface 5289void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5290{ 5291 mRsmpInIndex += buffer->frameCount; 5292 buffer->frameCount = 0; 5293} 5294 5295bool AudioFlinger::RecordThread::checkForNewParameters_l() 5296{ 5297 bool reconfig = false; 5298 5299 while (!mNewParameters.isEmpty()) { 5300 status_t status = NO_ERROR; 5301 String8 keyValuePair = mNewParameters[0]; 5302 AudioParameter param = AudioParameter(keyValuePair); 5303 int value; 5304 audio_format_t reqFormat = mFormat; 5305 int reqSamplingRate = mReqSampleRate; 5306 int reqChannelCount = mReqChannelCount; 5307 5308 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5309 reqSamplingRate = value; 5310 reconfig = true; 5311 } 5312 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5313 reqFormat = (audio_format_t) value; 5314 reconfig = true; 5315 } 5316 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5317 reqChannelCount = popcount(value); 5318 reconfig = true; 5319 } 5320 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5321 // do not accept frame count changes if tracks are open as the track buffer 5322 // size depends on frame count and correct behavior would not be guaranteed 5323 // if frame count is changed after track creation 5324 if (mActiveTrack != 0) { 5325 status = INVALID_OPERATION; 5326 } else { 5327 reconfig = true; 5328 } 5329 } 5330 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5331 // forward device change to effects that have requested to be 5332 // aware of attached audio device. 5333 for (size_t i = 0; i < mEffectChains.size(); i++) { 5334 mEffectChains[i]->setDevice_l(value); 5335 } 5336 // store input device and output device but do not forward output device to audio HAL. 5337 // Note that status is ignored by the caller for output device 5338 // (see AudioFlinger::setParameters() 5339 if (value & AUDIO_DEVICE_OUT_ALL) { 5340 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5341 status = BAD_VALUE; 5342 } else { 5343 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5344 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5345 if (mTrack != NULL) { 5346 bool suspend = audio_is_bluetooth_sco_device( 5347 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5348 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5349 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5350 } 5351 } 5352 mDevice |= (uint32_t)value; 5353 } 5354 if (status == NO_ERROR) { 5355 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5356 if (status == INVALID_OPERATION) { 5357 mInput->stream->common.standby(&mInput->stream->common); 5358 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5359 } 5360 if (reconfig) { 5361 if (status == BAD_VALUE && 5362 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5363 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5364 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5365 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5366 (reqChannelCount < 3)) { 5367 status = NO_ERROR; 5368 } 5369 if (status == NO_ERROR) { 5370 readInputParameters(); 5371 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5372 } 5373 } 5374 } 5375 5376 mNewParameters.removeAt(0); 5377 5378 mParamStatus = status; 5379 mParamCond.signal(); 5380 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5381 // already timed out waiting for the status and will never signal the condition. 5382 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5383 } 5384 return reconfig; 5385} 5386 5387String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5388{ 5389 char *s; 5390 String8 out_s8 = String8(); 5391 5392 Mutex::Autolock _l(mLock); 5393 if (initCheck() != NO_ERROR) { 5394 return out_s8; 5395 } 5396 5397 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5398 out_s8 = String8(s); 5399 free(s); 5400 return out_s8; 5401} 5402 5403void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5404 AudioSystem::OutputDescriptor desc; 5405 void *param2 = NULL; 5406 5407 switch (event) { 5408 case AudioSystem::INPUT_OPENED: 5409 case AudioSystem::INPUT_CONFIG_CHANGED: 5410 desc.channels = mChannelMask; 5411 desc.samplingRate = mSampleRate; 5412 desc.format = mFormat; 5413 desc.frameCount = mFrameCount; 5414 desc.latency = 0; 5415 param2 = &desc; 5416 break; 5417 5418 case AudioSystem::INPUT_CLOSED: 5419 default: 5420 break; 5421 } 5422 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5423} 5424 5425void AudioFlinger::RecordThread::readInputParameters() 5426{ 5427 delete mRsmpInBuffer; 5428 // mRsmpInBuffer is always assigned a new[] below 5429 delete mRsmpOutBuffer; 5430 mRsmpOutBuffer = NULL; 5431 delete mResampler; 5432 mResampler = NULL; 5433 5434 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5435 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5436 mChannelCount = (uint16_t)popcount(mChannelMask); 5437 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5438 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5439 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5440 mFrameCount = mInputBytes / mFrameSize; 5441 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5442 5443 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5444 { 5445 int channelCount; 5446 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5447 // stereo to mono post process as the resampler always outputs stereo. 5448 if (mChannelCount == 1 && mReqChannelCount == 2) { 5449 channelCount = 1; 5450 } else { 5451 channelCount = 2; 5452 } 5453 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5454 mResampler->setSampleRate(mSampleRate); 5455 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5456 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5457 5458 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5459 if (mChannelCount == 1 && mReqChannelCount == 1) { 5460 mFrameCount >>= 1; 5461 } 5462 5463 } 5464 mRsmpInIndex = mFrameCount; 5465} 5466 5467unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5468{ 5469 Mutex::Autolock _l(mLock); 5470 if (initCheck() != NO_ERROR) { 5471 return 0; 5472 } 5473 5474 return mInput->stream->get_input_frames_lost(mInput->stream); 5475} 5476 5477uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5478{ 5479 Mutex::Autolock _l(mLock); 5480 uint32_t result = 0; 5481 if (getEffectChain_l(sessionId) != 0) { 5482 result = EFFECT_SESSION; 5483 } 5484 5485 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5486 result |= TRACK_SESSION; 5487 } 5488 5489 return result; 5490} 5491 5492AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5493{ 5494 Mutex::Autolock _l(mLock); 5495 return mTrack; 5496} 5497 5498AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5499{ 5500 Mutex::Autolock _l(mLock); 5501 return mInput; 5502} 5503 5504AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5505{ 5506 Mutex::Autolock _l(mLock); 5507 AudioStreamIn *input = mInput; 5508 mInput = NULL; 5509 return input; 5510} 5511 5512// this method must always be called either with ThreadBase mLock held or inside the thread loop 5513audio_stream_t* AudioFlinger::RecordThread::stream() 5514{ 5515 if (mInput == NULL) { 5516 return NULL; 5517 } 5518 return &mInput->stream->common; 5519} 5520 5521 5522// ---------------------------------------------------------------------------- 5523 5524audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5525 uint32_t *pSamplingRate, 5526 audio_format_t *pFormat, 5527 uint32_t *pChannels, 5528 uint32_t *pLatencyMs, 5529 uint32_t flags) 5530{ 5531 status_t status; 5532 PlaybackThread *thread = NULL; 5533 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5534 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5535 uint32_t channels = pChannels ? *pChannels : 0; 5536 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5537 audio_stream_out_t *outStream; 5538 audio_hw_device_t *outHwDev; 5539 5540 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5541 pDevices ? *pDevices : 0, 5542 samplingRate, 5543 format, 5544 channels, 5545 flags); 5546 5547 if (pDevices == NULL || *pDevices == 0) { 5548 return 0; 5549 } 5550 5551 Mutex::Autolock _l(mLock); 5552 5553 outHwDev = findSuitableHwDev_l(*pDevices); 5554 if (outHwDev == NULL) 5555 return 0; 5556 5557 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5558 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5559 &channels, &samplingRate, &outStream); 5560 mHardwareStatus = AUDIO_HW_IDLE; 5561 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5562 outStream, 5563 samplingRate, 5564 format, 5565 channels, 5566 status); 5567 5568 if (outStream != NULL) { 5569 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5570 audio_io_handle_t id = nextUniqueId(); 5571 5572 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5573 (format != AUDIO_FORMAT_PCM_16_BIT) || 5574 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5575 thread = new DirectOutputThread(this, output, id, *pDevices); 5576 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5577 } else { 5578 thread = new MixerThread(this, output, id, *pDevices); 5579 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5580 } 5581 mPlaybackThreads.add(id, thread); 5582 5583 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5584 if (pFormat != NULL) *pFormat = format; 5585 if (pChannels != NULL) *pChannels = channels; 5586 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5587 5588 // notify client processes of the new output creation 5589 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5590 return id; 5591 } 5592 5593 return 0; 5594} 5595 5596audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5597 audio_io_handle_t output2) 5598{ 5599 Mutex::Autolock _l(mLock); 5600 MixerThread *thread1 = checkMixerThread_l(output1); 5601 MixerThread *thread2 = checkMixerThread_l(output2); 5602 5603 if (thread1 == NULL || thread2 == NULL) { 5604 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5605 return 0; 5606 } 5607 5608 audio_io_handle_t id = nextUniqueId(); 5609 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5610 thread->addOutputTrack(thread2); 5611 mPlaybackThreads.add(id, thread); 5612 // notify client processes of the new output creation 5613 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5614 return id; 5615} 5616 5617status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5618{ 5619 // keep strong reference on the playback thread so that 5620 // it is not destroyed while exit() is executed 5621 sp <PlaybackThread> thread; 5622 { 5623 Mutex::Autolock _l(mLock); 5624 thread = checkPlaybackThread_l(output); 5625 if (thread == NULL) { 5626 return BAD_VALUE; 5627 } 5628 5629 ALOGV("closeOutput() %d", output); 5630 5631 if (thread->type() == ThreadBase::MIXER) { 5632 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5633 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5634 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5635 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5636 } 5637 } 5638 } 5639 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5640 mPlaybackThreads.removeItem(output); 5641 } 5642 thread->exit(); 5643 // The thread entity (active unit of execution) is no longer running here, 5644 // but the ThreadBase container still exists. 5645 5646 if (thread->type() != ThreadBase::DUPLICATING) { 5647 AudioStreamOut *out = thread->clearOutput(); 5648 assert(out != NULL); 5649 // from now on thread->mOutput is NULL 5650 out->hwDev->close_output_stream(out->hwDev, out->stream); 5651 delete out; 5652 } 5653 return NO_ERROR; 5654} 5655 5656status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5657{ 5658 Mutex::Autolock _l(mLock); 5659 PlaybackThread *thread = checkPlaybackThread_l(output); 5660 5661 if (thread == NULL) { 5662 return BAD_VALUE; 5663 } 5664 5665 ALOGV("suspendOutput() %d", output); 5666 thread->suspend(); 5667 5668 return NO_ERROR; 5669} 5670 5671status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5672{ 5673 Mutex::Autolock _l(mLock); 5674 PlaybackThread *thread = checkPlaybackThread_l(output); 5675 5676 if (thread == NULL) { 5677 return BAD_VALUE; 5678 } 5679 5680 ALOGV("restoreOutput() %d", output); 5681 5682 thread->restore(); 5683 5684 return NO_ERROR; 5685} 5686 5687audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5688 uint32_t *pSamplingRate, 5689 audio_format_t *pFormat, 5690 uint32_t *pChannels, 5691 audio_in_acoustics_t acoustics) 5692{ 5693 status_t status; 5694 RecordThread *thread = NULL; 5695 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5696 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5697 uint32_t channels = pChannels ? *pChannels : 0; 5698 uint32_t reqSamplingRate = samplingRate; 5699 audio_format_t reqFormat = format; 5700 uint32_t reqChannels = channels; 5701 audio_stream_in_t *inStream; 5702 audio_hw_device_t *inHwDev; 5703 5704 if (pDevices == NULL || *pDevices == 0) { 5705 return 0; 5706 } 5707 5708 Mutex::Autolock _l(mLock); 5709 5710 inHwDev = findSuitableHwDev_l(*pDevices); 5711 if (inHwDev == NULL) 5712 return 0; 5713 5714 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5715 &channels, &samplingRate, 5716 acoustics, 5717 &inStream); 5718 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5719 inStream, 5720 samplingRate, 5721 format, 5722 channels, 5723 acoustics, 5724 status); 5725 5726 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5727 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5728 // or stereo to mono conversions on 16 bit PCM inputs. 5729 if (inStream == NULL && status == BAD_VALUE && 5730 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5731 (samplingRate <= 2 * reqSamplingRate) && 5732 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5733 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5734 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5735 &channels, &samplingRate, 5736 acoustics, 5737 &inStream); 5738 } 5739 5740 if (inStream != NULL) { 5741 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5742 5743 audio_io_handle_t id = nextUniqueId(); 5744 // Start record thread 5745 // RecorThread require both input and output device indication to forward to audio 5746 // pre processing modules 5747 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5748 thread = new RecordThread(this, 5749 input, 5750 reqSamplingRate, 5751 reqChannels, 5752 id, 5753 device); 5754 mRecordThreads.add(id, thread); 5755 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5756 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5757 if (pFormat != NULL) *pFormat = format; 5758 if (pChannels != NULL) *pChannels = reqChannels; 5759 5760 input->stream->common.standby(&input->stream->common); 5761 5762 // notify client processes of the new input creation 5763 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5764 return id; 5765 } 5766 5767 return 0; 5768} 5769 5770status_t AudioFlinger::closeInput(audio_io_handle_t input) 5771{ 5772 // keep strong reference on the record thread so that 5773 // it is not destroyed while exit() is executed 5774 sp <RecordThread> thread; 5775 { 5776 Mutex::Autolock _l(mLock); 5777 thread = checkRecordThread_l(input); 5778 if (thread == NULL) { 5779 return BAD_VALUE; 5780 } 5781 5782 ALOGV("closeInput() %d", input); 5783 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5784 mRecordThreads.removeItem(input); 5785 } 5786 thread->exit(); 5787 // The thread entity (active unit of execution) is no longer running here, 5788 // but the ThreadBase container still exists. 5789 5790 AudioStreamIn *in = thread->clearInput(); 5791 assert(in != NULL); 5792 // from now on thread->mInput is NULL 5793 in->hwDev->close_input_stream(in->hwDev, in->stream); 5794 delete in; 5795 5796 return NO_ERROR; 5797} 5798 5799status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5800{ 5801 Mutex::Autolock _l(mLock); 5802 MixerThread *dstThread = checkMixerThread_l(output); 5803 if (dstThread == NULL) { 5804 ALOGW("setStreamOutput() bad output id %d", output); 5805 return BAD_VALUE; 5806 } 5807 5808 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5809 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5810 5811 dstThread->setStreamValid(stream, true); 5812 5813 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5814 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5815 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5816 MixerThread *srcThread = (MixerThread *)thread; 5817 srcThread->setStreamValid(stream, false); 5818 srcThread->invalidateTracks(stream); 5819 } 5820 } 5821 5822 return NO_ERROR; 5823} 5824 5825 5826int AudioFlinger::newAudioSessionId() 5827{ 5828 return nextUniqueId(); 5829} 5830 5831void AudioFlinger::acquireAudioSessionId(int audioSession) 5832{ 5833 Mutex::Autolock _l(mLock); 5834 pid_t caller = IPCThreadState::self()->getCallingPid(); 5835 ALOGV("acquiring %d from %d", audioSession, caller); 5836 size_t num = mAudioSessionRefs.size(); 5837 for (size_t i = 0; i< num; i++) { 5838 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5839 if (ref->sessionid == audioSession && ref->pid == caller) { 5840 ref->cnt++; 5841 ALOGV(" incremented refcount to %d", ref->cnt); 5842 return; 5843 } 5844 } 5845 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5846 ALOGV(" added new entry for %d", audioSession); 5847} 5848 5849void AudioFlinger::releaseAudioSessionId(int audioSession) 5850{ 5851 Mutex::Autolock _l(mLock); 5852 pid_t caller = IPCThreadState::self()->getCallingPid(); 5853 ALOGV("releasing %d from %d", audioSession, caller); 5854 size_t num = mAudioSessionRefs.size(); 5855 for (size_t i = 0; i< num; i++) { 5856 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5857 if (ref->sessionid == audioSession && ref->pid == caller) { 5858 ref->cnt--; 5859 ALOGV(" decremented refcount to %d", ref->cnt); 5860 if (ref->cnt == 0) { 5861 mAudioSessionRefs.removeAt(i); 5862 delete ref; 5863 purgeStaleEffects_l(); 5864 } 5865 return; 5866 } 5867 } 5868 ALOGW("session id %d not found for pid %d", audioSession, caller); 5869} 5870 5871void AudioFlinger::purgeStaleEffects_l() { 5872 5873 ALOGV("purging stale effects"); 5874 5875 Vector< sp<EffectChain> > chains; 5876 5877 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5878 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5879 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5880 sp<EffectChain> ec = t->mEffectChains[j]; 5881 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5882 chains.push(ec); 5883 } 5884 } 5885 } 5886 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5887 sp<RecordThread> t = mRecordThreads.valueAt(i); 5888 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5889 sp<EffectChain> ec = t->mEffectChains[j]; 5890 chains.push(ec); 5891 } 5892 } 5893 5894 for (size_t i = 0; i < chains.size(); i++) { 5895 sp<EffectChain> ec = chains[i]; 5896 int sessionid = ec->sessionId(); 5897 sp<ThreadBase> t = ec->mThread.promote(); 5898 if (t == 0) { 5899 continue; 5900 } 5901 size_t numsessionrefs = mAudioSessionRefs.size(); 5902 bool found = false; 5903 for (size_t k = 0; k < numsessionrefs; k++) { 5904 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5905 if (ref->sessionid == sessionid) { 5906 ALOGV(" session %d still exists for %d with %d refs", 5907 sessionid, ref->pid, ref->cnt); 5908 found = true; 5909 break; 5910 } 5911 } 5912 if (!found) { 5913 // remove all effects from the chain 5914 while (ec->mEffects.size()) { 5915 sp<EffectModule> effect = ec->mEffects[0]; 5916 effect->unPin(); 5917 Mutex::Autolock _l (t->mLock); 5918 t->removeEffect_l(effect); 5919 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5920 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5921 if (handle != 0) { 5922 handle->mEffect.clear(); 5923 if (handle->mHasControl && handle->mEnabled) { 5924 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5925 } 5926 } 5927 } 5928 AudioSystem::unregisterEffect(effect->id()); 5929 } 5930 } 5931 } 5932 return; 5933} 5934 5935// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5936AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5937{ 5938 return mPlaybackThreads.valueFor(output).get(); 5939} 5940 5941// checkMixerThread_l() must be called with AudioFlinger::mLock held 5942AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5943{ 5944 PlaybackThread *thread = checkPlaybackThread_l(output); 5945 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5946} 5947 5948// checkRecordThread_l() must be called with AudioFlinger::mLock held 5949AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5950{ 5951 return mRecordThreads.valueFor(input).get(); 5952} 5953 5954uint32_t AudioFlinger::nextUniqueId() 5955{ 5956 return android_atomic_inc(&mNextUniqueId); 5957} 5958 5959AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5960{ 5961 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5962 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5963 AudioStreamOut *output = thread->getOutput(); 5964 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5965 return thread; 5966 } 5967 } 5968 return NULL; 5969} 5970 5971uint32_t AudioFlinger::primaryOutputDevice_l() const 5972{ 5973 PlaybackThread *thread = primaryPlaybackThread_l(); 5974 5975 if (thread == NULL) { 5976 return 0; 5977 } 5978 5979 return thread->device(); 5980} 5981 5982 5983// ---------------------------------------------------------------------------- 5984// Effect management 5985// ---------------------------------------------------------------------------- 5986 5987 5988status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5989{ 5990 Mutex::Autolock _l(mLock); 5991 return EffectQueryNumberEffects(numEffects); 5992} 5993 5994status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5995{ 5996 Mutex::Autolock _l(mLock); 5997 return EffectQueryEffect(index, descriptor); 5998} 5999 6000status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6001 effect_descriptor_t *descriptor) const 6002{ 6003 Mutex::Autolock _l(mLock); 6004 return EffectGetDescriptor(pUuid, descriptor); 6005} 6006 6007 6008sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6009 effect_descriptor_t *pDesc, 6010 const sp<IEffectClient>& effectClient, 6011 int32_t priority, 6012 audio_io_handle_t io, 6013 int sessionId, 6014 status_t *status, 6015 int *id, 6016 int *enabled) 6017{ 6018 status_t lStatus = NO_ERROR; 6019 sp<EffectHandle> handle; 6020 effect_descriptor_t desc; 6021 6022 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6023 pid, effectClient.get(), priority, sessionId, io); 6024 6025 if (pDesc == NULL) { 6026 lStatus = BAD_VALUE; 6027 goto Exit; 6028 } 6029 6030 // check audio settings permission for global effects 6031 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6032 lStatus = PERMISSION_DENIED; 6033 goto Exit; 6034 } 6035 6036 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6037 // that can only be created by audio policy manager (running in same process) 6038 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6039 lStatus = PERMISSION_DENIED; 6040 goto Exit; 6041 } 6042 6043 if (io == 0) { 6044 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6045 // output must be specified by AudioPolicyManager when using session 6046 // AUDIO_SESSION_OUTPUT_STAGE 6047 lStatus = BAD_VALUE; 6048 goto Exit; 6049 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6050 // if the output returned by getOutputForEffect() is removed before we lock the 6051 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6052 // and we will exit safely 6053 io = AudioSystem::getOutputForEffect(&desc); 6054 } 6055 } 6056 6057 { 6058 Mutex::Autolock _l(mLock); 6059 6060 6061 if (!EffectIsNullUuid(&pDesc->uuid)) { 6062 // if uuid is specified, request effect descriptor 6063 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6064 if (lStatus < 0) { 6065 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6066 goto Exit; 6067 } 6068 } else { 6069 // if uuid is not specified, look for an available implementation 6070 // of the required type in effect factory 6071 if (EffectIsNullUuid(&pDesc->type)) { 6072 ALOGW("createEffect() no effect type"); 6073 lStatus = BAD_VALUE; 6074 goto Exit; 6075 } 6076 uint32_t numEffects = 0; 6077 effect_descriptor_t d; 6078 d.flags = 0; // prevent compiler warning 6079 bool found = false; 6080 6081 lStatus = EffectQueryNumberEffects(&numEffects); 6082 if (lStatus < 0) { 6083 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6084 goto Exit; 6085 } 6086 for (uint32_t i = 0; i < numEffects; i++) { 6087 lStatus = EffectQueryEffect(i, &desc); 6088 if (lStatus < 0) { 6089 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6090 continue; 6091 } 6092 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6093 // If matching type found save effect descriptor. If the session is 6094 // 0 and the effect is not auxiliary, continue enumeration in case 6095 // an auxiliary version of this effect type is available 6096 found = true; 6097 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6098 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6099 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6100 break; 6101 } 6102 } 6103 } 6104 if (!found) { 6105 lStatus = BAD_VALUE; 6106 ALOGW("createEffect() effect not found"); 6107 goto Exit; 6108 } 6109 // For same effect type, chose auxiliary version over insert version if 6110 // connect to output mix (Compliance to OpenSL ES) 6111 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6112 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6113 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6114 } 6115 } 6116 6117 // Do not allow auxiliary effects on a session different from 0 (output mix) 6118 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6119 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6120 lStatus = INVALID_OPERATION; 6121 goto Exit; 6122 } 6123 6124 // check recording permission for visualizer 6125 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6126 !recordingAllowed()) { 6127 lStatus = PERMISSION_DENIED; 6128 goto Exit; 6129 } 6130 6131 // return effect descriptor 6132 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6133 6134 // If output is not specified try to find a matching audio session ID in one of the 6135 // output threads. 6136 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6137 // because of code checking output when entering the function. 6138 // Note: io is never 0 when creating an effect on an input 6139 if (io == 0) { 6140 // look for the thread where the specified audio session is present 6141 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6142 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6143 io = mPlaybackThreads.keyAt(i); 6144 break; 6145 } 6146 } 6147 if (io == 0) { 6148 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6149 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6150 io = mRecordThreads.keyAt(i); 6151 break; 6152 } 6153 } 6154 } 6155 // If no output thread contains the requested session ID, default to 6156 // first output. The effect chain will be moved to the correct output 6157 // thread when a track with the same session ID is created 6158 if (io == 0 && mPlaybackThreads.size()) { 6159 io = mPlaybackThreads.keyAt(0); 6160 } 6161 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6162 } 6163 ThreadBase *thread = checkRecordThread_l(io); 6164 if (thread == NULL) { 6165 thread = checkPlaybackThread_l(io); 6166 if (thread == NULL) { 6167 ALOGE("createEffect() unknown output thread"); 6168 lStatus = BAD_VALUE; 6169 goto Exit; 6170 } 6171 } 6172 6173 sp<Client> client = registerPid_l(pid); 6174 6175 // create effect on selected output thread 6176 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6177 &desc, enabled, &lStatus); 6178 if (handle != 0 && id != NULL) { 6179 *id = handle->id(); 6180 } 6181 } 6182 6183Exit: 6184 if(status) { 6185 *status = lStatus; 6186 } 6187 return handle; 6188} 6189 6190status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6191 audio_io_handle_t dstOutput) 6192{ 6193 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6194 sessionId, srcOutput, dstOutput); 6195 Mutex::Autolock _l(mLock); 6196 if (srcOutput == dstOutput) { 6197 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6198 return NO_ERROR; 6199 } 6200 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6201 if (srcThread == NULL) { 6202 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6203 return BAD_VALUE; 6204 } 6205 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6206 if (dstThread == NULL) { 6207 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6208 return BAD_VALUE; 6209 } 6210 6211 Mutex::Autolock _dl(dstThread->mLock); 6212 Mutex::Autolock _sl(srcThread->mLock); 6213 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6214 6215 return NO_ERROR; 6216} 6217 6218// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6219status_t AudioFlinger::moveEffectChain_l(int sessionId, 6220 AudioFlinger::PlaybackThread *srcThread, 6221 AudioFlinger::PlaybackThread *dstThread, 6222 bool reRegister) 6223{ 6224 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6225 sessionId, srcThread, dstThread); 6226 6227 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6228 if (chain == 0) { 6229 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6230 sessionId, srcThread); 6231 return INVALID_OPERATION; 6232 } 6233 6234 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6235 // so that a new chain is created with correct parameters when first effect is added. This is 6236 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6237 // removed. 6238 srcThread->removeEffectChain_l(chain); 6239 6240 // transfer all effects one by one so that new effect chain is created on new thread with 6241 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6242 audio_io_handle_t dstOutput = dstThread->id(); 6243 sp<EffectChain> dstChain; 6244 uint32_t strategy = 0; // prevent compiler warning 6245 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6246 while (effect != 0) { 6247 srcThread->removeEffect_l(effect); 6248 dstThread->addEffect_l(effect); 6249 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6250 if (effect->state() == EffectModule::ACTIVE || 6251 effect->state() == EffectModule::STOPPING) { 6252 effect->start(); 6253 } 6254 // if the move request is not received from audio policy manager, the effect must be 6255 // re-registered with the new strategy and output 6256 if (dstChain == 0) { 6257 dstChain = effect->chain().promote(); 6258 if (dstChain == 0) { 6259 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6260 srcThread->addEffect_l(effect); 6261 return NO_INIT; 6262 } 6263 strategy = dstChain->strategy(); 6264 } 6265 if (reRegister) { 6266 AudioSystem::unregisterEffect(effect->id()); 6267 AudioSystem::registerEffect(&effect->desc(), 6268 dstOutput, 6269 strategy, 6270 sessionId, 6271 effect->id()); 6272 } 6273 effect = chain->getEffectFromId_l(0); 6274 } 6275 6276 return NO_ERROR; 6277} 6278 6279 6280// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6281sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6282 const sp<AudioFlinger::Client>& client, 6283 const sp<IEffectClient>& effectClient, 6284 int32_t priority, 6285 int sessionId, 6286 effect_descriptor_t *desc, 6287 int *enabled, 6288 status_t *status 6289 ) 6290{ 6291 sp<EffectModule> effect; 6292 sp<EffectHandle> handle; 6293 status_t lStatus; 6294 sp<EffectChain> chain; 6295 bool chainCreated = false; 6296 bool effectCreated = false; 6297 bool effectRegistered = false; 6298 6299 lStatus = initCheck(); 6300 if (lStatus != NO_ERROR) { 6301 ALOGW("createEffect_l() Audio driver not initialized."); 6302 goto Exit; 6303 } 6304 6305 // Do not allow effects with session ID 0 on direct output or duplicating threads 6306 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6307 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6308 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6309 desc->name, sessionId); 6310 lStatus = BAD_VALUE; 6311 goto Exit; 6312 } 6313 // Only Pre processor effects are allowed on input threads and only on input threads 6314 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6315 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6316 desc->name, desc->flags, mType); 6317 lStatus = BAD_VALUE; 6318 goto Exit; 6319 } 6320 6321 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6322 6323 { // scope for mLock 6324 Mutex::Autolock _l(mLock); 6325 6326 // check for existing effect chain with the requested audio session 6327 chain = getEffectChain_l(sessionId); 6328 if (chain == 0) { 6329 // create a new chain for this session 6330 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6331 chain = new EffectChain(this, sessionId); 6332 addEffectChain_l(chain); 6333 chain->setStrategy(getStrategyForSession_l(sessionId)); 6334 chainCreated = true; 6335 } else { 6336 effect = chain->getEffectFromDesc_l(desc); 6337 } 6338 6339 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6340 6341 if (effect == 0) { 6342 int id = mAudioFlinger->nextUniqueId(); 6343 // Check CPU and memory usage 6344 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6345 if (lStatus != NO_ERROR) { 6346 goto Exit; 6347 } 6348 effectRegistered = true; 6349 // create a new effect module if none present in the chain 6350 effect = new EffectModule(this, chain, desc, id, sessionId); 6351 lStatus = effect->status(); 6352 if (lStatus != NO_ERROR) { 6353 goto Exit; 6354 } 6355 lStatus = chain->addEffect_l(effect); 6356 if (lStatus != NO_ERROR) { 6357 goto Exit; 6358 } 6359 effectCreated = true; 6360 6361 effect->setDevice(mDevice); 6362 effect->setMode(mAudioFlinger->getMode()); 6363 } 6364 // create effect handle and connect it to effect module 6365 handle = new EffectHandle(effect, client, effectClient, priority); 6366 lStatus = effect->addHandle(handle); 6367 if (enabled != NULL) { 6368 *enabled = (int)effect->isEnabled(); 6369 } 6370 } 6371 6372Exit: 6373 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6374 Mutex::Autolock _l(mLock); 6375 if (effectCreated) { 6376 chain->removeEffect_l(effect); 6377 } 6378 if (effectRegistered) { 6379 AudioSystem::unregisterEffect(effect->id()); 6380 } 6381 if (chainCreated) { 6382 removeEffectChain_l(chain); 6383 } 6384 handle.clear(); 6385 } 6386 6387 if(status) { 6388 *status = lStatus; 6389 } 6390 return handle; 6391} 6392 6393sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6394{ 6395 sp<EffectChain> chain = getEffectChain_l(sessionId); 6396 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6397} 6398 6399// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6400// PlaybackThread::mLock held 6401status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6402{ 6403 // check for existing effect chain with the requested audio session 6404 int sessionId = effect->sessionId(); 6405 sp<EffectChain> chain = getEffectChain_l(sessionId); 6406 bool chainCreated = false; 6407 6408 if (chain == 0) { 6409 // create a new chain for this session 6410 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6411 chain = new EffectChain(this, sessionId); 6412 addEffectChain_l(chain); 6413 chain->setStrategy(getStrategyForSession_l(sessionId)); 6414 chainCreated = true; 6415 } 6416 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6417 6418 if (chain->getEffectFromId_l(effect->id()) != 0) { 6419 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6420 this, effect->desc().name, chain.get()); 6421 return BAD_VALUE; 6422 } 6423 6424 status_t status = chain->addEffect_l(effect); 6425 if (status != NO_ERROR) { 6426 if (chainCreated) { 6427 removeEffectChain_l(chain); 6428 } 6429 return status; 6430 } 6431 6432 effect->setDevice(mDevice); 6433 effect->setMode(mAudioFlinger->getMode()); 6434 return NO_ERROR; 6435} 6436 6437void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6438 6439 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6440 effect_descriptor_t desc = effect->desc(); 6441 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6442 detachAuxEffect_l(effect->id()); 6443 } 6444 6445 sp<EffectChain> chain = effect->chain().promote(); 6446 if (chain != 0) { 6447 // remove effect chain if removing last effect 6448 if (chain->removeEffect_l(effect) == 0) { 6449 removeEffectChain_l(chain); 6450 } 6451 } else { 6452 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6453 } 6454} 6455 6456void AudioFlinger::ThreadBase::lockEffectChains_l( 6457 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6458{ 6459 effectChains = mEffectChains; 6460 for (size_t i = 0; i < mEffectChains.size(); i++) { 6461 mEffectChains[i]->lock(); 6462 } 6463} 6464 6465void AudioFlinger::ThreadBase::unlockEffectChains( 6466 const Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6467{ 6468 for (size_t i = 0; i < effectChains.size(); i++) { 6469 effectChains[i]->unlock(); 6470 } 6471} 6472 6473sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6474{ 6475 Mutex::Autolock _l(mLock); 6476 return getEffectChain_l(sessionId); 6477} 6478 6479sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6480{ 6481 size_t size = mEffectChains.size(); 6482 for (size_t i = 0; i < size; i++) { 6483 if (mEffectChains[i]->sessionId() == sessionId) { 6484 return mEffectChains[i]; 6485 } 6486 } 6487 return 0; 6488} 6489 6490void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6491{ 6492 Mutex::Autolock _l(mLock); 6493 size_t size = mEffectChains.size(); 6494 for (size_t i = 0; i < size; i++) { 6495 mEffectChains[i]->setMode_l(mode); 6496 } 6497} 6498 6499void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6500 const wp<EffectHandle>& handle, 6501 bool unpinIfLast) { 6502 6503 Mutex::Autolock _l(mLock); 6504 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6505 // delete the effect module if removing last handle on it 6506 if (effect->removeHandle(handle) == 0) { 6507 if (!effect->isPinned() || unpinIfLast) { 6508 removeEffect_l(effect); 6509 AudioSystem::unregisterEffect(effect->id()); 6510 } 6511 } 6512} 6513 6514status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6515{ 6516 int session = chain->sessionId(); 6517 int16_t *buffer = mMixBuffer; 6518 bool ownsBuffer = false; 6519 6520 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6521 if (session > 0) { 6522 // Only one effect chain can be present in direct output thread and it uses 6523 // the mix buffer as input 6524 if (mType != DIRECT) { 6525 size_t numSamples = mFrameCount * mChannelCount; 6526 buffer = new int16_t[numSamples]; 6527 memset(buffer, 0, numSamples * sizeof(int16_t)); 6528 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6529 ownsBuffer = true; 6530 } 6531 6532 // Attach all tracks with same session ID to this chain. 6533 for (size_t i = 0; i < mTracks.size(); ++i) { 6534 sp<Track> track = mTracks[i]; 6535 if (session == track->sessionId()) { 6536 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6537 track->setMainBuffer(buffer); 6538 chain->incTrackCnt(); 6539 } 6540 } 6541 6542 // indicate all active tracks in the chain 6543 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6544 sp<Track> track = mActiveTracks[i].promote(); 6545 if (track == 0) continue; 6546 if (session == track->sessionId()) { 6547 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6548 chain->incActiveTrackCnt(); 6549 } 6550 } 6551 } 6552 6553 chain->setInBuffer(buffer, ownsBuffer); 6554 chain->setOutBuffer(mMixBuffer); 6555 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6556 // chains list in order to be processed last as it contains output stage effects 6557 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6558 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6559 // after track specific effects and before output stage 6560 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6561 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6562 // Effect chain for other sessions are inserted at beginning of effect 6563 // chains list to be processed before output mix effects. Relative order between other 6564 // sessions is not important 6565 size_t size = mEffectChains.size(); 6566 size_t i = 0; 6567 for (i = 0; i < size; i++) { 6568 if (mEffectChains[i]->sessionId() < session) break; 6569 } 6570 mEffectChains.insertAt(chain, i); 6571 checkSuspendOnAddEffectChain_l(chain); 6572 6573 return NO_ERROR; 6574} 6575 6576size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6577{ 6578 int session = chain->sessionId(); 6579 6580 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6581 6582 for (size_t i = 0; i < mEffectChains.size(); i++) { 6583 if (chain == mEffectChains[i]) { 6584 mEffectChains.removeAt(i); 6585 // detach all active tracks from the chain 6586 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6587 sp<Track> track = mActiveTracks[i].promote(); 6588 if (track == 0) continue; 6589 if (session == track->sessionId()) { 6590 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6591 chain.get(), session); 6592 chain->decActiveTrackCnt(); 6593 } 6594 } 6595 6596 // detach all tracks with same session ID from this chain 6597 for (size_t i = 0; i < mTracks.size(); ++i) { 6598 sp<Track> track = mTracks[i]; 6599 if (session == track->sessionId()) { 6600 track->setMainBuffer(mMixBuffer); 6601 chain->decTrackCnt(); 6602 } 6603 } 6604 break; 6605 } 6606 } 6607 return mEffectChains.size(); 6608} 6609 6610status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6611 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6612{ 6613 Mutex::Autolock _l(mLock); 6614 return attachAuxEffect_l(track, EffectId); 6615} 6616 6617status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6618 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6619{ 6620 status_t status = NO_ERROR; 6621 6622 if (EffectId == 0) { 6623 track->setAuxBuffer(0, NULL); 6624 } else { 6625 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6626 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6627 if (effect != 0) { 6628 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6629 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6630 } else { 6631 status = INVALID_OPERATION; 6632 } 6633 } else { 6634 status = BAD_VALUE; 6635 } 6636 } 6637 return status; 6638} 6639 6640void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6641{ 6642 for (size_t i = 0; i < mTracks.size(); ++i) { 6643 sp<Track> track = mTracks[i]; 6644 if (track->auxEffectId() == effectId) { 6645 attachAuxEffect_l(track, 0); 6646 } 6647 } 6648} 6649 6650status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6651{ 6652 // only one chain per input thread 6653 if (mEffectChains.size() != 0) { 6654 return INVALID_OPERATION; 6655 } 6656 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6657 6658 chain->setInBuffer(NULL); 6659 chain->setOutBuffer(NULL); 6660 6661 checkSuspendOnAddEffectChain_l(chain); 6662 6663 mEffectChains.add(chain); 6664 6665 return NO_ERROR; 6666} 6667 6668size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6669{ 6670 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6671 ALOGW_IF(mEffectChains.size() != 1, 6672 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6673 chain.get(), mEffectChains.size(), this); 6674 if (mEffectChains.size() == 1) { 6675 mEffectChains.removeAt(0); 6676 } 6677 return 0; 6678} 6679 6680// ---------------------------------------------------------------------------- 6681// EffectModule implementation 6682// ---------------------------------------------------------------------------- 6683 6684#undef LOG_TAG 6685#define LOG_TAG "AudioFlinger::EffectModule" 6686 6687AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6688 const wp<AudioFlinger::EffectChain>& chain, 6689 effect_descriptor_t *desc, 6690 int id, 6691 int sessionId) 6692 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6693 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6694{ 6695 ALOGV("Constructor %p", this); 6696 int lStatus; 6697 if (thread == NULL) { 6698 return; 6699 } 6700 6701 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6702 6703 // create effect engine from effect factory 6704 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6705 6706 if (mStatus != NO_ERROR) { 6707 return; 6708 } 6709 lStatus = init(); 6710 if (lStatus < 0) { 6711 mStatus = lStatus; 6712 goto Error; 6713 } 6714 6715 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6716 mPinned = true; 6717 } 6718 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6719 return; 6720Error: 6721 EffectRelease(mEffectInterface); 6722 mEffectInterface = NULL; 6723 ALOGV("Constructor Error %d", mStatus); 6724} 6725 6726AudioFlinger::EffectModule::~EffectModule() 6727{ 6728 ALOGV("Destructor %p", this); 6729 if (mEffectInterface != NULL) { 6730 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6731 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6732 sp<ThreadBase> thread = mThread.promote(); 6733 if (thread != 0) { 6734 audio_stream_t *stream = thread->stream(); 6735 if (stream != NULL) { 6736 stream->remove_audio_effect(stream, mEffectInterface); 6737 } 6738 } 6739 } 6740 // release effect engine 6741 EffectRelease(mEffectInterface); 6742 } 6743} 6744 6745status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6746{ 6747 status_t status; 6748 6749 Mutex::Autolock _l(mLock); 6750 int priority = handle->priority(); 6751 size_t size = mHandles.size(); 6752 sp<EffectHandle> h; 6753 size_t i; 6754 for (i = 0; i < size; i++) { 6755 h = mHandles[i].promote(); 6756 if (h == 0) continue; 6757 if (h->priority() <= priority) break; 6758 } 6759 // if inserted in first place, move effect control from previous owner to this handle 6760 if (i == 0) { 6761 bool enabled = false; 6762 if (h != 0) { 6763 enabled = h->enabled(); 6764 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6765 } 6766 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6767 status = NO_ERROR; 6768 } else { 6769 status = ALREADY_EXISTS; 6770 } 6771 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6772 mHandles.insertAt(handle, i); 6773 return status; 6774} 6775 6776size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6777{ 6778 Mutex::Autolock _l(mLock); 6779 size_t size = mHandles.size(); 6780 size_t i; 6781 for (i = 0; i < size; i++) { 6782 if (mHandles[i] == handle) break; 6783 } 6784 if (i == size) { 6785 return size; 6786 } 6787 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6788 6789 bool enabled = false; 6790 EffectHandle *hdl = handle.unsafe_get(); 6791 if (hdl != NULL) { 6792 ALOGV("removeHandle() unsafe_get OK"); 6793 enabled = hdl->enabled(); 6794 } 6795 mHandles.removeAt(i); 6796 size = mHandles.size(); 6797 // if removed from first place, move effect control from this handle to next in line 6798 if (i == 0 && size != 0) { 6799 sp<EffectHandle> h = mHandles[0].promote(); 6800 if (h != 0) { 6801 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6802 } 6803 } 6804 6805 // Prevent calls to process() and other functions on effect interface from now on. 6806 // The effect engine will be released by the destructor when the last strong reference on 6807 // this object is released which can happen after next process is called. 6808 if (size == 0 && !mPinned) { 6809 mState = DESTROYED; 6810 } 6811 6812 return size; 6813} 6814 6815sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6816{ 6817 Mutex::Autolock _l(mLock); 6818 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6819} 6820 6821void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6822{ 6823 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6824 // keep a strong reference on this EffectModule to avoid calling the 6825 // destructor before we exit 6826 sp<EffectModule> keep(this); 6827 { 6828 sp<ThreadBase> thread = mThread.promote(); 6829 if (thread != 0) { 6830 thread->disconnectEffect(keep, handle, unpinIfLast); 6831 } 6832 } 6833} 6834 6835void AudioFlinger::EffectModule::updateState() { 6836 Mutex::Autolock _l(mLock); 6837 6838 switch (mState) { 6839 case RESTART: 6840 reset_l(); 6841 // FALL THROUGH 6842 6843 case STARTING: 6844 // clear auxiliary effect input buffer for next accumulation 6845 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6846 memset(mConfig.inputCfg.buffer.raw, 6847 0, 6848 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6849 } 6850 start_l(); 6851 mState = ACTIVE; 6852 break; 6853 case STOPPING: 6854 stop_l(); 6855 mDisableWaitCnt = mMaxDisableWaitCnt; 6856 mState = STOPPED; 6857 break; 6858 case STOPPED: 6859 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6860 // turn off sequence. 6861 if (--mDisableWaitCnt == 0) { 6862 reset_l(); 6863 mState = IDLE; 6864 } 6865 break; 6866 default: //IDLE , ACTIVE, DESTROYED 6867 break; 6868 } 6869} 6870 6871void AudioFlinger::EffectModule::process() 6872{ 6873 Mutex::Autolock _l(mLock); 6874 6875 if (mState == DESTROYED || mEffectInterface == NULL || 6876 mConfig.inputCfg.buffer.raw == NULL || 6877 mConfig.outputCfg.buffer.raw == NULL) { 6878 return; 6879 } 6880 6881 if (isProcessEnabled()) { 6882 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6883 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6884 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6885 mConfig.inputCfg.buffer.s32, 6886 mConfig.inputCfg.buffer.frameCount/2); 6887 } 6888 6889 // do the actual processing in the effect engine 6890 int ret = (*mEffectInterface)->process(mEffectInterface, 6891 &mConfig.inputCfg.buffer, 6892 &mConfig.outputCfg.buffer); 6893 6894 // force transition to IDLE state when engine is ready 6895 if (mState == STOPPED && ret == -ENODATA) { 6896 mDisableWaitCnt = 1; 6897 } 6898 6899 // clear auxiliary effect input buffer for next accumulation 6900 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6901 memset(mConfig.inputCfg.buffer.raw, 0, 6902 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6903 } 6904 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6905 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6906 // If an insert effect is idle and input buffer is different from output buffer, 6907 // accumulate input onto output 6908 sp<EffectChain> chain = mChain.promote(); 6909 if (chain != 0 && chain->activeTrackCnt() != 0) { 6910 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6911 int16_t *in = mConfig.inputCfg.buffer.s16; 6912 int16_t *out = mConfig.outputCfg.buffer.s16; 6913 for (size_t i = 0; i < frameCnt; i++) { 6914 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6915 } 6916 } 6917 } 6918} 6919 6920void AudioFlinger::EffectModule::reset_l() 6921{ 6922 if (mEffectInterface == NULL) { 6923 return; 6924 } 6925 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6926} 6927 6928status_t AudioFlinger::EffectModule::configure() 6929{ 6930 uint32_t channels; 6931 if (mEffectInterface == NULL) { 6932 return NO_INIT; 6933 } 6934 6935 sp<ThreadBase> thread = mThread.promote(); 6936 if (thread == 0) { 6937 return DEAD_OBJECT; 6938 } 6939 6940 // TODO: handle configuration of effects replacing track process 6941 if (thread->channelCount() == 1) { 6942 channels = AUDIO_CHANNEL_OUT_MONO; 6943 } else { 6944 channels = AUDIO_CHANNEL_OUT_STEREO; 6945 } 6946 6947 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6948 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6949 } else { 6950 mConfig.inputCfg.channels = channels; 6951 } 6952 mConfig.outputCfg.channels = channels; 6953 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6954 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6955 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6956 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6957 mConfig.inputCfg.bufferProvider.cookie = NULL; 6958 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6959 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6960 mConfig.outputCfg.bufferProvider.cookie = NULL; 6961 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6962 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6963 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6964 // Insert effect: 6965 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6966 // always overwrites output buffer: input buffer == output buffer 6967 // - in other sessions: 6968 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6969 // other effect: overwrites output buffer: input buffer == output buffer 6970 // Auxiliary effect: 6971 // accumulates in output buffer: input buffer != output buffer 6972 // Therefore: accumulate <=> input buffer != output buffer 6973 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6974 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6975 } else { 6976 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6977 } 6978 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6979 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6980 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6981 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6982 6983 ALOGV("configure() %p thread %p buffer %p framecount %d", 6984 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6985 6986 status_t cmdStatus; 6987 uint32_t size = sizeof(int); 6988 status_t status = (*mEffectInterface)->command(mEffectInterface, 6989 EFFECT_CMD_SET_CONFIG, 6990 sizeof(effect_config_t), 6991 &mConfig, 6992 &size, 6993 &cmdStatus); 6994 if (status == 0) { 6995 status = cmdStatus; 6996 } 6997 6998 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6999 (1000 * mConfig.outputCfg.buffer.frameCount); 7000 7001 return status; 7002} 7003 7004status_t AudioFlinger::EffectModule::init() 7005{ 7006 Mutex::Autolock _l(mLock); 7007 if (mEffectInterface == NULL) { 7008 return NO_INIT; 7009 } 7010 status_t cmdStatus; 7011 uint32_t size = sizeof(status_t); 7012 status_t status = (*mEffectInterface)->command(mEffectInterface, 7013 EFFECT_CMD_INIT, 7014 0, 7015 NULL, 7016 &size, 7017 &cmdStatus); 7018 if (status == 0) { 7019 status = cmdStatus; 7020 } 7021 return status; 7022} 7023 7024status_t AudioFlinger::EffectModule::start() 7025{ 7026 Mutex::Autolock _l(mLock); 7027 return start_l(); 7028} 7029 7030status_t AudioFlinger::EffectModule::start_l() 7031{ 7032 if (mEffectInterface == NULL) { 7033 return NO_INIT; 7034 } 7035 status_t cmdStatus; 7036 uint32_t size = sizeof(status_t); 7037 status_t status = (*mEffectInterface)->command(mEffectInterface, 7038 EFFECT_CMD_ENABLE, 7039 0, 7040 NULL, 7041 &size, 7042 &cmdStatus); 7043 if (status == 0) { 7044 status = cmdStatus; 7045 } 7046 if (status == 0 && 7047 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7048 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7049 sp<ThreadBase> thread = mThread.promote(); 7050 if (thread != 0) { 7051 audio_stream_t *stream = thread->stream(); 7052 if (stream != NULL) { 7053 stream->add_audio_effect(stream, mEffectInterface); 7054 } 7055 } 7056 } 7057 return status; 7058} 7059 7060status_t AudioFlinger::EffectModule::stop() 7061{ 7062 Mutex::Autolock _l(mLock); 7063 return stop_l(); 7064} 7065 7066status_t AudioFlinger::EffectModule::stop_l() 7067{ 7068 if (mEffectInterface == NULL) { 7069 return NO_INIT; 7070 } 7071 status_t cmdStatus; 7072 uint32_t size = sizeof(status_t); 7073 status_t status = (*mEffectInterface)->command(mEffectInterface, 7074 EFFECT_CMD_DISABLE, 7075 0, 7076 NULL, 7077 &size, 7078 &cmdStatus); 7079 if (status == 0) { 7080 status = cmdStatus; 7081 } 7082 if (status == 0 && 7083 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7084 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7085 sp<ThreadBase> thread = mThread.promote(); 7086 if (thread != 0) { 7087 audio_stream_t *stream = thread->stream(); 7088 if (stream != NULL) { 7089 stream->remove_audio_effect(stream, mEffectInterface); 7090 } 7091 } 7092 } 7093 return status; 7094} 7095 7096status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7097 uint32_t cmdSize, 7098 void *pCmdData, 7099 uint32_t *replySize, 7100 void *pReplyData) 7101{ 7102 Mutex::Autolock _l(mLock); 7103// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7104 7105 if (mState == DESTROYED || mEffectInterface == NULL) { 7106 return NO_INIT; 7107 } 7108 status_t status = (*mEffectInterface)->command(mEffectInterface, 7109 cmdCode, 7110 cmdSize, 7111 pCmdData, 7112 replySize, 7113 pReplyData); 7114 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7115 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7116 for (size_t i = 1; i < mHandles.size(); i++) { 7117 sp<EffectHandle> h = mHandles[i].promote(); 7118 if (h != 0) { 7119 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7120 } 7121 } 7122 } 7123 return status; 7124} 7125 7126status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7127{ 7128 7129 Mutex::Autolock _l(mLock); 7130 ALOGV("setEnabled %p enabled %d", this, enabled); 7131 7132 if (enabled != isEnabled()) { 7133 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7134 if (enabled && status != NO_ERROR) { 7135 return status; 7136 } 7137 7138 switch (mState) { 7139 // going from disabled to enabled 7140 case IDLE: 7141 mState = STARTING; 7142 break; 7143 case STOPPED: 7144 mState = RESTART; 7145 break; 7146 case STOPPING: 7147 mState = ACTIVE; 7148 break; 7149 7150 // going from enabled to disabled 7151 case RESTART: 7152 mState = STOPPED; 7153 break; 7154 case STARTING: 7155 mState = IDLE; 7156 break; 7157 case ACTIVE: 7158 mState = STOPPING; 7159 break; 7160 case DESTROYED: 7161 return NO_ERROR; // simply ignore as we are being destroyed 7162 } 7163 for (size_t i = 1; i < mHandles.size(); i++) { 7164 sp<EffectHandle> h = mHandles[i].promote(); 7165 if (h != 0) { 7166 h->setEnabled(enabled); 7167 } 7168 } 7169 } 7170 return NO_ERROR; 7171} 7172 7173bool AudioFlinger::EffectModule::isEnabled() const 7174{ 7175 switch (mState) { 7176 case RESTART: 7177 case STARTING: 7178 case ACTIVE: 7179 return true; 7180 case IDLE: 7181 case STOPPING: 7182 case STOPPED: 7183 case DESTROYED: 7184 default: 7185 return false; 7186 } 7187} 7188 7189bool AudioFlinger::EffectModule::isProcessEnabled() const 7190{ 7191 switch (mState) { 7192 case RESTART: 7193 case ACTIVE: 7194 case STOPPING: 7195 case STOPPED: 7196 return true; 7197 case IDLE: 7198 case STARTING: 7199 case DESTROYED: 7200 default: 7201 return false; 7202 } 7203} 7204 7205status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7206{ 7207 Mutex::Autolock _l(mLock); 7208 status_t status = NO_ERROR; 7209 7210 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7211 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7212 if (isProcessEnabled() && 7213 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7214 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7215 status_t cmdStatus; 7216 uint32_t volume[2]; 7217 uint32_t *pVolume = NULL; 7218 uint32_t size = sizeof(volume); 7219 volume[0] = *left; 7220 volume[1] = *right; 7221 if (controller) { 7222 pVolume = volume; 7223 } 7224 status = (*mEffectInterface)->command(mEffectInterface, 7225 EFFECT_CMD_SET_VOLUME, 7226 size, 7227 volume, 7228 &size, 7229 pVolume); 7230 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7231 *left = volume[0]; 7232 *right = volume[1]; 7233 } 7234 } 7235 return status; 7236} 7237 7238status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7239{ 7240 Mutex::Autolock _l(mLock); 7241 status_t status = NO_ERROR; 7242 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7243 // audio pre processing modules on RecordThread can receive both output and 7244 // input device indication in the same call 7245 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7246 if (dev) { 7247 status_t cmdStatus; 7248 uint32_t size = sizeof(status_t); 7249 7250 status = (*mEffectInterface)->command(mEffectInterface, 7251 EFFECT_CMD_SET_DEVICE, 7252 sizeof(uint32_t), 7253 &dev, 7254 &size, 7255 &cmdStatus); 7256 if (status == NO_ERROR) { 7257 status = cmdStatus; 7258 } 7259 } 7260 dev = device & AUDIO_DEVICE_IN_ALL; 7261 if (dev) { 7262 status_t cmdStatus; 7263 uint32_t size = sizeof(status_t); 7264 7265 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7266 EFFECT_CMD_SET_INPUT_DEVICE, 7267 sizeof(uint32_t), 7268 &dev, 7269 &size, 7270 &cmdStatus); 7271 if (status2 == NO_ERROR) { 7272 status2 = cmdStatus; 7273 } 7274 if (status == NO_ERROR) { 7275 status = status2; 7276 } 7277 } 7278 } 7279 return status; 7280} 7281 7282status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7283{ 7284 Mutex::Autolock _l(mLock); 7285 status_t status = NO_ERROR; 7286 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7287 status_t cmdStatus; 7288 uint32_t size = sizeof(status_t); 7289 status = (*mEffectInterface)->command(mEffectInterface, 7290 EFFECT_CMD_SET_AUDIO_MODE, 7291 sizeof(audio_mode_t), 7292 &mode, 7293 &size, 7294 &cmdStatus); 7295 if (status == NO_ERROR) { 7296 status = cmdStatus; 7297 } 7298 } 7299 return status; 7300} 7301 7302void AudioFlinger::EffectModule::setSuspended(bool suspended) 7303{ 7304 Mutex::Autolock _l(mLock); 7305 mSuspended = suspended; 7306} 7307 7308bool AudioFlinger::EffectModule::suspended() const 7309{ 7310 Mutex::Autolock _l(mLock); 7311 return mSuspended; 7312} 7313 7314status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7315{ 7316 const size_t SIZE = 256; 7317 char buffer[SIZE]; 7318 String8 result; 7319 7320 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7321 result.append(buffer); 7322 7323 bool locked = tryLock(mLock); 7324 // failed to lock - AudioFlinger is probably deadlocked 7325 if (!locked) { 7326 result.append("\t\tCould not lock Fx mutex:\n"); 7327 } 7328 7329 result.append("\t\tSession Status State Engine:\n"); 7330 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7331 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7332 result.append(buffer); 7333 7334 result.append("\t\tDescriptor:\n"); 7335 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7336 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7337 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7338 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7339 result.append(buffer); 7340 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7341 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7342 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7343 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7344 result.append(buffer); 7345 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7346 mDescriptor.apiVersion, 7347 mDescriptor.flags); 7348 result.append(buffer); 7349 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7350 mDescriptor.name); 7351 result.append(buffer); 7352 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7353 mDescriptor.implementor); 7354 result.append(buffer); 7355 7356 result.append("\t\t- Input configuration:\n"); 7357 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7358 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7359 (uint32_t)mConfig.inputCfg.buffer.raw, 7360 mConfig.inputCfg.buffer.frameCount, 7361 mConfig.inputCfg.samplingRate, 7362 mConfig.inputCfg.channels, 7363 mConfig.inputCfg.format); 7364 result.append(buffer); 7365 7366 result.append("\t\t- Output configuration:\n"); 7367 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7368 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7369 (uint32_t)mConfig.outputCfg.buffer.raw, 7370 mConfig.outputCfg.buffer.frameCount, 7371 mConfig.outputCfg.samplingRate, 7372 mConfig.outputCfg.channels, 7373 mConfig.outputCfg.format); 7374 result.append(buffer); 7375 7376 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7377 result.append(buffer); 7378 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7379 for (size_t i = 0; i < mHandles.size(); ++i) { 7380 sp<EffectHandle> handle = mHandles[i].promote(); 7381 if (handle != 0) { 7382 handle->dump(buffer, SIZE); 7383 result.append(buffer); 7384 } 7385 } 7386 7387 result.append("\n"); 7388 7389 write(fd, result.string(), result.length()); 7390 7391 if (locked) { 7392 mLock.unlock(); 7393 } 7394 7395 return NO_ERROR; 7396} 7397 7398// ---------------------------------------------------------------------------- 7399// EffectHandle implementation 7400// ---------------------------------------------------------------------------- 7401 7402#undef LOG_TAG 7403#define LOG_TAG "AudioFlinger::EffectHandle" 7404 7405AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7406 const sp<AudioFlinger::Client>& client, 7407 const sp<IEffectClient>& effectClient, 7408 int32_t priority) 7409 : BnEffect(), 7410 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7411 mPriority(priority), mHasControl(false), mEnabled(false) 7412{ 7413 ALOGV("constructor %p", this); 7414 7415 if (client == 0) { 7416 return; 7417 } 7418 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7419 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7420 if (mCblkMemory != 0) { 7421 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7422 7423 if (mCblk != NULL) { 7424 new(mCblk) effect_param_cblk_t(); 7425 mBuffer = (uint8_t *)mCblk + bufOffset; 7426 } 7427 } else { 7428 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7429 return; 7430 } 7431} 7432 7433AudioFlinger::EffectHandle::~EffectHandle() 7434{ 7435 ALOGV("Destructor %p", this); 7436 disconnect(false); 7437 ALOGV("Destructor DONE %p", this); 7438} 7439 7440status_t AudioFlinger::EffectHandle::enable() 7441{ 7442 ALOGV("enable %p", this); 7443 if (!mHasControl) return INVALID_OPERATION; 7444 if (mEffect == 0) return DEAD_OBJECT; 7445 7446 if (mEnabled) { 7447 return NO_ERROR; 7448 } 7449 7450 mEnabled = true; 7451 7452 sp<ThreadBase> thread = mEffect->thread().promote(); 7453 if (thread != 0) { 7454 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7455 } 7456 7457 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7458 if (mEffect->suspended()) { 7459 return NO_ERROR; 7460 } 7461 7462 status_t status = mEffect->setEnabled(true); 7463 if (status != NO_ERROR) { 7464 if (thread != 0) { 7465 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7466 } 7467 mEnabled = false; 7468 } 7469 return status; 7470} 7471 7472status_t AudioFlinger::EffectHandle::disable() 7473{ 7474 ALOGV("disable %p", this); 7475 if (!mHasControl) return INVALID_OPERATION; 7476 if (mEffect == 0) return DEAD_OBJECT; 7477 7478 if (!mEnabled) { 7479 return NO_ERROR; 7480 } 7481 mEnabled = false; 7482 7483 if (mEffect->suspended()) { 7484 return NO_ERROR; 7485 } 7486 7487 status_t status = mEffect->setEnabled(false); 7488 7489 sp<ThreadBase> thread = mEffect->thread().promote(); 7490 if (thread != 0) { 7491 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7492 } 7493 7494 return status; 7495} 7496 7497void AudioFlinger::EffectHandle::disconnect() 7498{ 7499 disconnect(true); 7500} 7501 7502void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7503{ 7504 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7505 if (mEffect == 0) { 7506 return; 7507 } 7508 mEffect->disconnect(this, unpinIfLast); 7509 7510 if (mHasControl && mEnabled) { 7511 sp<ThreadBase> thread = mEffect->thread().promote(); 7512 if (thread != 0) { 7513 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7514 } 7515 } 7516 7517 // release sp on module => module destructor can be called now 7518 mEffect.clear(); 7519 if (mClient != 0) { 7520 if (mCblk != NULL) { 7521 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7522 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7523 } 7524 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7525 // Client destructor must run with AudioFlinger mutex locked 7526 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7527 mClient.clear(); 7528 } 7529} 7530 7531status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7532 uint32_t cmdSize, 7533 void *pCmdData, 7534 uint32_t *replySize, 7535 void *pReplyData) 7536{ 7537// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7538// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7539 7540 // only get parameter command is permitted for applications not controlling the effect 7541 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7542 return INVALID_OPERATION; 7543 } 7544 if (mEffect == 0) return DEAD_OBJECT; 7545 if (mClient == 0) return INVALID_OPERATION; 7546 7547 // handle commands that are not forwarded transparently to effect engine 7548 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7549 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7550 // no risk to block the whole media server process or mixer threads is we are stuck here 7551 Mutex::Autolock _l(mCblk->lock); 7552 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7553 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7554 mCblk->serverIndex = 0; 7555 mCblk->clientIndex = 0; 7556 return BAD_VALUE; 7557 } 7558 status_t status = NO_ERROR; 7559 while (mCblk->serverIndex < mCblk->clientIndex) { 7560 int reply; 7561 uint32_t rsize = sizeof(int); 7562 int *p = (int *)(mBuffer + mCblk->serverIndex); 7563 int size = *p++; 7564 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7565 ALOGW("command(): invalid parameter block size"); 7566 break; 7567 } 7568 effect_param_t *param = (effect_param_t *)p; 7569 if (param->psize == 0 || param->vsize == 0) { 7570 ALOGW("command(): null parameter or value size"); 7571 mCblk->serverIndex += size; 7572 continue; 7573 } 7574 uint32_t psize = sizeof(effect_param_t) + 7575 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7576 param->vsize; 7577 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7578 psize, 7579 p, 7580 &rsize, 7581 &reply); 7582 // stop at first error encountered 7583 if (ret != NO_ERROR) { 7584 status = ret; 7585 *(int *)pReplyData = reply; 7586 break; 7587 } else if (reply != NO_ERROR) { 7588 *(int *)pReplyData = reply; 7589 break; 7590 } 7591 mCblk->serverIndex += size; 7592 } 7593 mCblk->serverIndex = 0; 7594 mCblk->clientIndex = 0; 7595 return status; 7596 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7597 *(int *)pReplyData = NO_ERROR; 7598 return enable(); 7599 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7600 *(int *)pReplyData = NO_ERROR; 7601 return disable(); 7602 } 7603 7604 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7605} 7606 7607void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7608{ 7609 ALOGV("setControl %p control %d", this, hasControl); 7610 7611 mHasControl = hasControl; 7612 mEnabled = enabled; 7613 7614 if (signal && mEffectClient != 0) { 7615 mEffectClient->controlStatusChanged(hasControl); 7616 } 7617} 7618 7619void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7620 uint32_t cmdSize, 7621 void *pCmdData, 7622 uint32_t replySize, 7623 void *pReplyData) 7624{ 7625 if (mEffectClient != 0) { 7626 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7627 } 7628} 7629 7630 7631 7632void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7633{ 7634 if (mEffectClient != 0) { 7635 mEffectClient->enableStatusChanged(enabled); 7636 } 7637} 7638 7639status_t AudioFlinger::EffectHandle::onTransact( 7640 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7641{ 7642 return BnEffect::onTransact(code, data, reply, flags); 7643} 7644 7645 7646void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7647{ 7648 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7649 7650 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7651 (mClient == 0) ? getpid_cached : mClient->pid(), 7652 mPriority, 7653 mHasControl, 7654 !locked, 7655 mCblk ? mCblk->clientIndex : 0, 7656 mCblk ? mCblk->serverIndex : 0 7657 ); 7658 7659 if (locked) { 7660 mCblk->lock.unlock(); 7661 } 7662} 7663 7664#undef LOG_TAG 7665#define LOG_TAG "AudioFlinger::EffectChain" 7666 7667AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7668 int sessionId) 7669 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7670 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7671 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7672{ 7673 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7674 if (thread == NULL) { 7675 return; 7676 } 7677 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7678 thread->frameCount(); 7679} 7680 7681AudioFlinger::EffectChain::~EffectChain() 7682{ 7683 if (mOwnInBuffer) { 7684 delete mInBuffer; 7685 } 7686 7687} 7688 7689// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7690sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7691{ 7692 size_t size = mEffects.size(); 7693 7694 for (size_t i = 0; i < size; i++) { 7695 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7696 return mEffects[i]; 7697 } 7698 } 7699 return 0; 7700} 7701 7702// getEffectFromId_l() must be called with ThreadBase::mLock held 7703sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7704{ 7705 size_t size = mEffects.size(); 7706 7707 for (size_t i = 0; i < size; i++) { 7708 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7709 if (id == 0 || mEffects[i]->id() == id) { 7710 return mEffects[i]; 7711 } 7712 } 7713 return 0; 7714} 7715 7716// getEffectFromType_l() must be called with ThreadBase::mLock held 7717sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7718 const effect_uuid_t *type) 7719{ 7720 size_t size = mEffects.size(); 7721 7722 for (size_t i = 0; i < size; i++) { 7723 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7724 return mEffects[i]; 7725 } 7726 } 7727 return 0; 7728} 7729 7730// Must be called with EffectChain::mLock locked 7731void AudioFlinger::EffectChain::process_l() 7732{ 7733 sp<ThreadBase> thread = mThread.promote(); 7734 if (thread == 0) { 7735 ALOGW("process_l(): cannot promote mixer thread"); 7736 return; 7737 } 7738 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7739 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7740 // always process effects unless no more tracks are on the session and the effect tail 7741 // has been rendered 7742 bool doProcess = true; 7743 if (!isGlobalSession) { 7744 bool tracksOnSession = (trackCnt() != 0); 7745 7746 if (!tracksOnSession && mTailBufferCount == 0) { 7747 doProcess = false; 7748 } 7749 7750 if (activeTrackCnt() == 0) { 7751 // if no track is active and the effect tail has not been rendered, 7752 // the input buffer must be cleared here as the mixer process will not do it 7753 if (tracksOnSession || mTailBufferCount > 0) { 7754 size_t numSamples = thread->frameCount() * thread->channelCount(); 7755 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7756 if (mTailBufferCount > 0) { 7757 mTailBufferCount--; 7758 } 7759 } 7760 } 7761 } 7762 7763 size_t size = mEffects.size(); 7764 if (doProcess) { 7765 for (size_t i = 0; i < size; i++) { 7766 mEffects[i]->process(); 7767 } 7768 } 7769 for (size_t i = 0; i < size; i++) { 7770 mEffects[i]->updateState(); 7771 } 7772} 7773 7774// addEffect_l() must be called with PlaybackThread::mLock held 7775status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7776{ 7777 effect_descriptor_t desc = effect->desc(); 7778 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7779 7780 Mutex::Autolock _l(mLock); 7781 effect->setChain(this); 7782 sp<ThreadBase> thread = mThread.promote(); 7783 if (thread == 0) { 7784 return NO_INIT; 7785 } 7786 effect->setThread(thread); 7787 7788 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7789 // Auxiliary effects are inserted at the beginning of mEffects vector as 7790 // they are processed first and accumulated in chain input buffer 7791 mEffects.insertAt(effect, 0); 7792 7793 // the input buffer for auxiliary effect contains mono samples in 7794 // 32 bit format. This is to avoid saturation in AudoMixer 7795 // accumulation stage. Saturation is done in EffectModule::process() before 7796 // calling the process in effect engine 7797 size_t numSamples = thread->frameCount(); 7798 int32_t *buffer = new int32_t[numSamples]; 7799 memset(buffer, 0, numSamples * sizeof(int32_t)); 7800 effect->setInBuffer((int16_t *)buffer); 7801 // auxiliary effects output samples to chain input buffer for further processing 7802 // by insert effects 7803 effect->setOutBuffer(mInBuffer); 7804 } else { 7805 // Insert effects are inserted at the end of mEffects vector as they are processed 7806 // after track and auxiliary effects. 7807 // Insert effect order as a function of indicated preference: 7808 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7809 // another effect is present 7810 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7811 // last effect claiming first position 7812 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7813 // first effect claiming last position 7814 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7815 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7816 // already present 7817 7818 size_t size = mEffects.size(); 7819 size_t idx_insert = size; 7820 ssize_t idx_insert_first = -1; 7821 ssize_t idx_insert_last = -1; 7822 7823 for (size_t i = 0; i < size; i++) { 7824 effect_descriptor_t d = mEffects[i]->desc(); 7825 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7826 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7827 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7828 // check invalid effect chaining combinations 7829 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7830 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7831 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7832 return INVALID_OPERATION; 7833 } 7834 // remember position of first insert effect and by default 7835 // select this as insert position for new effect 7836 if (idx_insert == size) { 7837 idx_insert = i; 7838 } 7839 // remember position of last insert effect claiming 7840 // first position 7841 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7842 idx_insert_first = i; 7843 } 7844 // remember position of first insert effect claiming 7845 // last position 7846 if (iPref == EFFECT_FLAG_INSERT_LAST && 7847 idx_insert_last == -1) { 7848 idx_insert_last = i; 7849 } 7850 } 7851 } 7852 7853 // modify idx_insert from first position if needed 7854 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7855 if (idx_insert_last != -1) { 7856 idx_insert = idx_insert_last; 7857 } else { 7858 idx_insert = size; 7859 } 7860 } else { 7861 if (idx_insert_first != -1) { 7862 idx_insert = idx_insert_first + 1; 7863 } 7864 } 7865 7866 // always read samples from chain input buffer 7867 effect->setInBuffer(mInBuffer); 7868 7869 // if last effect in the chain, output samples to chain 7870 // output buffer, otherwise to chain input buffer 7871 if (idx_insert == size) { 7872 if (idx_insert != 0) { 7873 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7874 mEffects[idx_insert-1]->configure(); 7875 } 7876 effect->setOutBuffer(mOutBuffer); 7877 } else { 7878 effect->setOutBuffer(mInBuffer); 7879 } 7880 mEffects.insertAt(effect, idx_insert); 7881 7882 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7883 } 7884 effect->configure(); 7885 return NO_ERROR; 7886} 7887 7888// removeEffect_l() must be called with PlaybackThread::mLock held 7889size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7890{ 7891 Mutex::Autolock _l(mLock); 7892 size_t size = mEffects.size(); 7893 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7894 7895 for (size_t i = 0; i < size; i++) { 7896 if (effect == mEffects[i]) { 7897 // calling stop here will remove pre-processing effect from the audio HAL. 7898 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7899 // the middle of a read from audio HAL 7900 if (mEffects[i]->state() == EffectModule::ACTIVE || 7901 mEffects[i]->state() == EffectModule::STOPPING) { 7902 mEffects[i]->stop(); 7903 } 7904 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7905 delete[] effect->inBuffer(); 7906 } else { 7907 if (i == size - 1 && i != 0) { 7908 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7909 mEffects[i - 1]->configure(); 7910 } 7911 } 7912 mEffects.removeAt(i); 7913 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7914 break; 7915 } 7916 } 7917 7918 return mEffects.size(); 7919} 7920 7921// setDevice_l() must be called with PlaybackThread::mLock held 7922void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7923{ 7924 size_t size = mEffects.size(); 7925 for (size_t i = 0; i < size; i++) { 7926 mEffects[i]->setDevice(device); 7927 } 7928} 7929 7930// setMode_l() must be called with PlaybackThread::mLock held 7931void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7932{ 7933 size_t size = mEffects.size(); 7934 for (size_t i = 0; i < size; i++) { 7935 mEffects[i]->setMode(mode); 7936 } 7937} 7938 7939// setVolume_l() must be called with PlaybackThread::mLock held 7940bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7941{ 7942 uint32_t newLeft = *left; 7943 uint32_t newRight = *right; 7944 bool hasControl = false; 7945 int ctrlIdx = -1; 7946 size_t size = mEffects.size(); 7947 7948 // first update volume controller 7949 for (size_t i = size; i > 0; i--) { 7950 if (mEffects[i - 1]->isProcessEnabled() && 7951 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7952 ctrlIdx = i - 1; 7953 hasControl = true; 7954 break; 7955 } 7956 } 7957 7958 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7959 if (hasControl) { 7960 *left = mNewLeftVolume; 7961 *right = mNewRightVolume; 7962 } 7963 return hasControl; 7964 } 7965 7966 mVolumeCtrlIdx = ctrlIdx; 7967 mLeftVolume = newLeft; 7968 mRightVolume = newRight; 7969 7970 // second get volume update from volume controller 7971 if (ctrlIdx >= 0) { 7972 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7973 mNewLeftVolume = newLeft; 7974 mNewRightVolume = newRight; 7975 } 7976 // then indicate volume to all other effects in chain. 7977 // Pass altered volume to effects before volume controller 7978 // and requested volume to effects after controller 7979 uint32_t lVol = newLeft; 7980 uint32_t rVol = newRight; 7981 7982 for (size_t i = 0; i < size; i++) { 7983 if ((int)i == ctrlIdx) continue; 7984 // this also works for ctrlIdx == -1 when there is no volume controller 7985 if ((int)i > ctrlIdx) { 7986 lVol = *left; 7987 rVol = *right; 7988 } 7989 mEffects[i]->setVolume(&lVol, &rVol, false); 7990 } 7991 *left = newLeft; 7992 *right = newRight; 7993 7994 return hasControl; 7995} 7996 7997status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7998{ 7999 const size_t SIZE = 256; 8000 char buffer[SIZE]; 8001 String8 result; 8002 8003 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8004 result.append(buffer); 8005 8006 bool locked = tryLock(mLock); 8007 // failed to lock - AudioFlinger is probably deadlocked 8008 if (!locked) { 8009 result.append("\tCould not lock mutex:\n"); 8010 } 8011 8012 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8013 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8014 mEffects.size(), 8015 (uint32_t)mInBuffer, 8016 (uint32_t)mOutBuffer, 8017 mActiveTrackCnt); 8018 result.append(buffer); 8019 write(fd, result.string(), result.size()); 8020 8021 for (size_t i = 0; i < mEffects.size(); ++i) { 8022 sp<EffectModule> effect = mEffects[i]; 8023 if (effect != 0) { 8024 effect->dump(fd, args); 8025 } 8026 } 8027 8028 if (locked) { 8029 mLock.unlock(); 8030 } 8031 8032 return NO_ERROR; 8033} 8034 8035// must be called with ThreadBase::mLock held 8036void AudioFlinger::EffectChain::setEffectSuspended_l( 8037 const effect_uuid_t *type, bool suspend) 8038{ 8039 sp<SuspendedEffectDesc> desc; 8040 // use effect type UUID timelow as key as there is no real risk of identical 8041 // timeLow fields among effect type UUIDs. 8042 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8043 if (suspend) { 8044 if (index >= 0) { 8045 desc = mSuspendedEffects.valueAt(index); 8046 } else { 8047 desc = new SuspendedEffectDesc(); 8048 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8049 mSuspendedEffects.add(type->timeLow, desc); 8050 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8051 } 8052 if (desc->mRefCount++ == 0) { 8053 sp<EffectModule> effect = getEffectIfEnabled(type); 8054 if (effect != 0) { 8055 desc->mEffect = effect; 8056 effect->setSuspended(true); 8057 effect->setEnabled(false); 8058 } 8059 } 8060 } else { 8061 if (index < 0) { 8062 return; 8063 } 8064 desc = mSuspendedEffects.valueAt(index); 8065 if (desc->mRefCount <= 0) { 8066 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8067 desc->mRefCount = 1; 8068 } 8069 if (--desc->mRefCount == 0) { 8070 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8071 if (desc->mEffect != 0) { 8072 sp<EffectModule> effect = desc->mEffect.promote(); 8073 if (effect != 0) { 8074 effect->setSuspended(false); 8075 sp<EffectHandle> handle = effect->controlHandle(); 8076 if (handle != 0) { 8077 effect->setEnabled(handle->enabled()); 8078 } 8079 } 8080 desc->mEffect.clear(); 8081 } 8082 mSuspendedEffects.removeItemsAt(index); 8083 } 8084 } 8085} 8086 8087// must be called with ThreadBase::mLock held 8088void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8089{ 8090 sp<SuspendedEffectDesc> desc; 8091 8092 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8093 if (suspend) { 8094 if (index >= 0) { 8095 desc = mSuspendedEffects.valueAt(index); 8096 } else { 8097 desc = new SuspendedEffectDesc(); 8098 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8099 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8100 } 8101 if (desc->mRefCount++ == 0) { 8102 Vector< sp<EffectModule> > effects; 8103 getSuspendEligibleEffects(effects); 8104 for (size_t i = 0; i < effects.size(); i++) { 8105 setEffectSuspended_l(&effects[i]->desc().type, true); 8106 } 8107 } 8108 } else { 8109 if (index < 0) { 8110 return; 8111 } 8112 desc = mSuspendedEffects.valueAt(index); 8113 if (desc->mRefCount <= 0) { 8114 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8115 desc->mRefCount = 1; 8116 } 8117 if (--desc->mRefCount == 0) { 8118 Vector<const effect_uuid_t *> types; 8119 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8120 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8121 continue; 8122 } 8123 types.add(&mSuspendedEffects.valueAt(i)->mType); 8124 } 8125 for (size_t i = 0; i < types.size(); i++) { 8126 setEffectSuspended_l(types[i], false); 8127 } 8128 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8129 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8130 } 8131 } 8132} 8133 8134 8135// The volume effect is used for automated tests only 8136#ifndef OPENSL_ES_H_ 8137static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8138 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8139const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8140#endif //OPENSL_ES_H_ 8141 8142bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8143{ 8144 // auxiliary effects and visualizer are never suspended on output mix 8145 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8146 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8147 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8148 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8149 return false; 8150 } 8151 return true; 8152} 8153 8154void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8155{ 8156 effects.clear(); 8157 for (size_t i = 0; i < mEffects.size(); i++) { 8158 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8159 effects.add(mEffects[i]); 8160 } 8161 } 8162} 8163 8164sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8165 const effect_uuid_t *type) 8166{ 8167 sp<EffectModule> effect = getEffectFromType_l(type); 8168 return effect != 0 && effect->isEnabled() ? effect : 0; 8169} 8170 8171void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8172 bool enabled) 8173{ 8174 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8175 if (enabled) { 8176 if (index < 0) { 8177 // if the effect is not suspend check if all effects are suspended 8178 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8179 if (index < 0) { 8180 return; 8181 } 8182 if (!isEffectEligibleForSuspend(effect->desc())) { 8183 return; 8184 } 8185 setEffectSuspended_l(&effect->desc().type, enabled); 8186 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8187 if (index < 0) { 8188 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8189 return; 8190 } 8191 } 8192 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8193 effect->desc().type.timeLow); 8194 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8195 // if effect is requested to suspended but was not yet enabled, supend it now. 8196 if (desc->mEffect == 0) { 8197 desc->mEffect = effect; 8198 effect->setEnabled(false); 8199 effect->setSuspended(true); 8200 } 8201 } else { 8202 if (index < 0) { 8203 return; 8204 } 8205 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8206 effect->desc().type.timeLow); 8207 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8208 desc->mEffect.clear(); 8209 effect->setSuspended(false); 8210 } 8211} 8212 8213#undef LOG_TAG 8214#define LOG_TAG "AudioFlinger" 8215 8216// ---------------------------------------------------------------------------- 8217 8218status_t AudioFlinger::onTransact( 8219 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8220{ 8221 return BnAudioFlinger::onTransact(code, data, reply, flags); 8222} 8223 8224}; // namespace android 8225