AudioFlinger.cpp revision e628d515888baadba75442128678e747e930ed58
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_INIT; 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_INIT; 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 status_t result; 827 828 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 829 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 830 // check calling permissions 831 if (!settingsAllowed()) { 832 return PERMISSION_DENIED; 833 } 834 835 // ioHandle == 0 means the parameters are global to the audio hardware interface 836 if (ioHandle == 0) { 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_SET_PARAMETER; 839 status_t final_result = NO_ERROR; 840 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 841 audio_hw_device_t *dev = mAudioHwDevs[i]; 842 result = dev->set_parameters(dev, keyValuePairs.string()); 843 final_result = result ?: final_result; 844 } 845 mHardwareStatus = AUDIO_HW_IDLE; 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 audio_hw_device_t *dev = mAudioHwDevs[i]; 909 char *s = dev->get_parameters(dev, keys.string()); 910 out_s8 += String8(s ? s : ""); 911 free(s); 912 } 913 return out_s8; 914 } 915 916 Mutex::Autolock _l(mLock); 917 918 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 919 if (playbackThread != NULL) { 920 return playbackThread->getParameters(keys); 921 } 922 RecordThread *recordThread = checkRecordThread_l(ioHandle); 923 if (recordThread != NULL) { 924 return recordThread->getParameters(keys); 925 } 926 return String8(""); 927} 928 929size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 930{ 931 status_t ret = initCheck(); 932 if (ret != NO_ERROR) { 933 return 0; 934 } 935 936 AutoMutex lock(mHardwareLock); 937 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 938 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 939 mHardwareStatus = AUDIO_HW_IDLE; 940 return size; 941} 942 943unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 944{ 945 if (ioHandle == 0) { 946 return 0; 947 } 948 949 Mutex::Autolock _l(mLock); 950 951 RecordThread *recordThread = checkRecordThread_l(ioHandle); 952 if (recordThread != NULL) { 953 return recordThread->getInputFramesLost(); 954 } 955 return 0; 956} 957 958status_t AudioFlinger::setVoiceVolume(float value) 959{ 960 status_t ret = initCheck(); 961 if (ret != NO_ERROR) { 962 return ret; 963 } 964 965 // check calling permissions 966 if (!settingsAllowed()) { 967 return PERMISSION_DENIED; 968 } 969 970 AutoMutex lock(mHardwareLock); 971 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 972 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 973 mHardwareStatus = AUDIO_HW_IDLE; 974 975 return ret; 976} 977 978status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 979 audio_io_handle_t output) const 980{ 981 status_t status; 982 983 Mutex::Autolock _l(mLock); 984 985 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 986 if (playbackThread != NULL) { 987 return playbackThread->getRenderPosition(halFrames, dspFrames); 988 } 989 990 return BAD_VALUE; 991} 992 993void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 994{ 995 996 Mutex::Autolock _l(mLock); 997 998 pid_t pid = IPCThreadState::self()->getCallingPid(); 999 if (mNotificationClients.indexOfKey(pid) < 0) { 1000 sp<NotificationClient> notificationClient = new NotificationClient(this, 1001 client, 1002 pid); 1003 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1004 1005 mNotificationClients.add(pid, notificationClient); 1006 1007 sp<IBinder> binder = client->asBinder(); 1008 binder->linkToDeath(notificationClient); 1009 1010 // the config change is always sent from playback or record threads to avoid deadlock 1011 // with AudioSystem::gLock 1012 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1013 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1014 } 1015 1016 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1017 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1018 } 1019 } 1020} 1021 1022void AudioFlinger::removeNotificationClient(pid_t pid) 1023{ 1024 Mutex::Autolock _l(mLock); 1025 1026 ssize_t index = mNotificationClients.indexOfKey(pid); 1027 if (index >= 0) { 1028 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 1029 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 1030 mNotificationClients.removeItem(pid); 1031 } 1032 1033 ALOGV("%d died, releasing its sessions", pid); 1034 size_t num = mAudioSessionRefs.size(); 1035 bool removed = false; 1036 for (size_t i = 0; i< num; ) { 1037 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1038 ALOGV(" pid %d @ %d", ref->pid, i); 1039 if (ref->pid == pid) { 1040 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1041 mAudioSessionRefs.removeAt(i); 1042 delete ref; 1043 removed = true; 1044 num--; 1045 } else { 1046 i++; 1047 } 1048 } 1049 if (removed) { 1050 purgeStaleEffects_l(); 1051 } 1052} 1053 1054// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1055void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1056{ 1057 size_t size = mNotificationClients.size(); 1058 for (size_t i = 0; i < size; i++) { 1059 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1060 param2); 1061 } 1062} 1063 1064// removeClient_l() must be called with AudioFlinger::mLock held 1065void AudioFlinger::removeClient_l(pid_t pid) 1066{ 1067 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1068 mClients.removeItem(pid); 1069} 1070 1071 1072// ---------------------------------------------------------------------------- 1073 1074AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1075 uint32_t device, type_t type) 1076 : Thread(false), 1077 mType(type), 1078 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1079 // mChannelMask 1080 mChannelCount(0), 1081 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1082 mParamStatus(NO_ERROR), 1083 mStandby(false), mId(id), 1084 mDevice(device), 1085 mDeathRecipient(new PMDeathRecipient(this)) 1086{ 1087} 1088 1089AudioFlinger::ThreadBase::~ThreadBase() 1090{ 1091 mParamCond.broadcast(); 1092 // do not lock the mutex in destructor 1093 releaseWakeLock_l(); 1094 if (mPowerManager != 0) { 1095 sp<IBinder> binder = mPowerManager->asBinder(); 1096 binder->unlinkToDeath(mDeathRecipient); 1097 } 1098} 1099 1100void AudioFlinger::ThreadBase::exit() 1101{ 1102 ALOGV("ThreadBase::exit"); 1103 { 1104 // This lock prevents the following race in thread (uniprocessor for illustration): 1105 // if (!exitPending()) { 1106 // // context switch from here to exit() 1107 // // exit() calls requestExit(), what exitPending() observes 1108 // // exit() calls signal(), which is dropped since no waiters 1109 // // context switch back from exit() to here 1110 // mWaitWorkCV.wait(...); 1111 // // now thread is hung 1112 // } 1113 AutoMutex lock(mLock); 1114 requestExit(); 1115 mWaitWorkCV.signal(); 1116 } 1117 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1118 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1119 requestExitAndWait(); 1120} 1121 1122status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1123{ 1124 status_t status; 1125 1126 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1127 Mutex::Autolock _l(mLock); 1128 1129 mNewParameters.add(keyValuePairs); 1130 mWaitWorkCV.signal(); 1131 // wait condition with timeout in case the thread loop has exited 1132 // before the request could be processed 1133 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1134 status = mParamStatus; 1135 mWaitWorkCV.signal(); 1136 } else { 1137 status = TIMED_OUT; 1138 } 1139 return status; 1140} 1141 1142void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1143{ 1144 Mutex::Autolock _l(mLock); 1145 sendConfigEvent_l(event, param); 1146} 1147 1148// sendConfigEvent_l() must be called with ThreadBase::mLock held 1149void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1150{ 1151 ConfigEvent configEvent; 1152 configEvent.mEvent = event; 1153 configEvent.mParam = param; 1154 mConfigEvents.add(configEvent); 1155 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1156 mWaitWorkCV.signal(); 1157} 1158 1159void AudioFlinger::ThreadBase::processConfigEvents() 1160{ 1161 mLock.lock(); 1162 while(!mConfigEvents.isEmpty()) { 1163 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1164 ConfigEvent configEvent = mConfigEvents[0]; 1165 mConfigEvents.removeAt(0); 1166 // release mLock before locking AudioFlinger mLock: lock order is always 1167 // AudioFlinger then ThreadBase to avoid cross deadlock 1168 mLock.unlock(); 1169 mAudioFlinger->mLock.lock(); 1170 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1171 mAudioFlinger->mLock.unlock(); 1172 mLock.lock(); 1173 } 1174 mLock.unlock(); 1175} 1176 1177status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1178{ 1179 const size_t SIZE = 256; 1180 char buffer[SIZE]; 1181 String8 result; 1182 1183 bool locked = tryLock(mLock); 1184 if (!locked) { 1185 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1186 write(fd, buffer, strlen(buffer)); 1187 } 1188 1189 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1190 result.append(buffer); 1191 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1192 result.append(buffer); 1193 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1194 result.append(buffer); 1195 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1196 result.append(buffer); 1197 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1198 result.append(buffer); 1199 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1200 result.append(buffer); 1201 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1202 result.append(buffer); 1203 1204 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1205 result.append(buffer); 1206 result.append(" Index Command"); 1207 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1208 snprintf(buffer, SIZE, "\n %02d ", i); 1209 result.append(buffer); 1210 result.append(mNewParameters[i]); 1211 } 1212 1213 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1214 result.append(buffer); 1215 snprintf(buffer, SIZE, " Index event param\n"); 1216 result.append(buffer); 1217 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1218 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1219 result.append(buffer); 1220 } 1221 result.append("\n"); 1222 1223 write(fd, result.string(), result.size()); 1224 1225 if (locked) { 1226 mLock.unlock(); 1227 } 1228 return NO_ERROR; 1229} 1230 1231status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1232{ 1233 const size_t SIZE = 256; 1234 char buffer[SIZE]; 1235 String8 result; 1236 1237 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1238 write(fd, buffer, strlen(buffer)); 1239 1240 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1241 sp<EffectChain> chain = mEffectChains[i]; 1242 if (chain != 0) { 1243 chain->dump(fd, args); 1244 } 1245 } 1246 return NO_ERROR; 1247} 1248 1249void AudioFlinger::ThreadBase::acquireWakeLock() 1250{ 1251 Mutex::Autolock _l(mLock); 1252 acquireWakeLock_l(); 1253} 1254 1255void AudioFlinger::ThreadBase::acquireWakeLock_l() 1256{ 1257 if (mPowerManager == 0) { 1258 // use checkService() to avoid blocking if power service is not up yet 1259 sp<IBinder> binder = 1260 defaultServiceManager()->checkService(String16("power")); 1261 if (binder == 0) { 1262 ALOGW("Thread %s cannot connect to the power manager service", mName); 1263 } else { 1264 mPowerManager = interface_cast<IPowerManager>(binder); 1265 binder->linkToDeath(mDeathRecipient); 1266 } 1267 } 1268 if (mPowerManager != 0) { 1269 sp<IBinder> binder = new BBinder(); 1270 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1271 binder, 1272 String16(mName)); 1273 if (status == NO_ERROR) { 1274 mWakeLockToken = binder; 1275 } 1276 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1277 } 1278} 1279 1280void AudioFlinger::ThreadBase::releaseWakeLock() 1281{ 1282 Mutex::Autolock _l(mLock); 1283 releaseWakeLock_l(); 1284} 1285 1286void AudioFlinger::ThreadBase::releaseWakeLock_l() 1287{ 1288 if (mWakeLockToken != 0) { 1289 ALOGV("releaseWakeLock_l() %s", mName); 1290 if (mPowerManager != 0) { 1291 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1292 } 1293 mWakeLockToken.clear(); 1294 } 1295} 1296 1297void AudioFlinger::ThreadBase::clearPowerManager() 1298{ 1299 Mutex::Autolock _l(mLock); 1300 releaseWakeLock_l(); 1301 mPowerManager.clear(); 1302} 1303 1304void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1305{ 1306 sp<ThreadBase> thread = mThread.promote(); 1307 if (thread != 0) { 1308 thread->clearPowerManager(); 1309 } 1310 ALOGW("power manager service died !!!"); 1311} 1312 1313void AudioFlinger::ThreadBase::setEffectSuspended( 1314 const effect_uuid_t *type, bool suspend, int sessionId) 1315{ 1316 Mutex::Autolock _l(mLock); 1317 setEffectSuspended_l(type, suspend, sessionId); 1318} 1319 1320void AudioFlinger::ThreadBase::setEffectSuspended_l( 1321 const effect_uuid_t *type, bool suspend, int sessionId) 1322{ 1323 sp<EffectChain> chain = getEffectChain_l(sessionId); 1324 if (chain != 0) { 1325 if (type != NULL) { 1326 chain->setEffectSuspended_l(type, suspend); 1327 } else { 1328 chain->setEffectSuspendedAll_l(suspend); 1329 } 1330 } 1331 1332 updateSuspendedSessions_l(type, suspend, sessionId); 1333} 1334 1335void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1336{ 1337 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1338 if (index < 0) { 1339 return; 1340 } 1341 1342 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1343 mSuspendedSessions.editValueAt(index); 1344 1345 for (size_t i = 0; i < sessionEffects.size(); i++) { 1346 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1347 for (int j = 0; j < desc->mRefCount; j++) { 1348 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1349 chain->setEffectSuspendedAll_l(true); 1350 } else { 1351 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1352 desc->mType.timeLow); 1353 chain->setEffectSuspended_l(&desc->mType, true); 1354 } 1355 } 1356 } 1357} 1358 1359void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1360 bool suspend, 1361 int sessionId) 1362{ 1363 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1364 1365 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1366 1367 if (suspend) { 1368 if (index >= 0) { 1369 sessionEffects = mSuspendedSessions.editValueAt(index); 1370 } else { 1371 mSuspendedSessions.add(sessionId, sessionEffects); 1372 } 1373 } else { 1374 if (index < 0) { 1375 return; 1376 } 1377 sessionEffects = mSuspendedSessions.editValueAt(index); 1378 } 1379 1380 1381 int key = EffectChain::kKeyForSuspendAll; 1382 if (type != NULL) { 1383 key = type->timeLow; 1384 } 1385 index = sessionEffects.indexOfKey(key); 1386 1387 sp <SuspendedSessionDesc> desc; 1388 if (suspend) { 1389 if (index >= 0) { 1390 desc = sessionEffects.valueAt(index); 1391 } else { 1392 desc = new SuspendedSessionDesc(); 1393 if (type != NULL) { 1394 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1395 } 1396 sessionEffects.add(key, desc); 1397 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1398 } 1399 desc->mRefCount++; 1400 } else { 1401 if (index < 0) { 1402 return; 1403 } 1404 desc = sessionEffects.valueAt(index); 1405 if (--desc->mRefCount == 0) { 1406 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1407 sessionEffects.removeItemsAt(index); 1408 if (sessionEffects.isEmpty()) { 1409 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1410 sessionId); 1411 mSuspendedSessions.removeItem(sessionId); 1412 } 1413 } 1414 } 1415 if (!sessionEffects.isEmpty()) { 1416 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1417 } 1418} 1419 1420void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1421 bool enabled, 1422 int sessionId) 1423{ 1424 Mutex::Autolock _l(mLock); 1425 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1426} 1427 1428void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1429 bool enabled, 1430 int sessionId) 1431{ 1432 if (mType != RECORD) { 1433 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1434 // another session. This gives the priority to well behaved effect control panels 1435 // and applications not using global effects. 1436 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1437 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1438 } 1439 } 1440 1441 sp<EffectChain> chain = getEffectChain_l(sessionId); 1442 if (chain != 0) { 1443 chain->checkSuspendOnEffectEnabled(effect, enabled); 1444 } 1445} 1446 1447// ---------------------------------------------------------------------------- 1448 1449AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1450 AudioStreamOut* output, 1451 audio_io_handle_t id, 1452 uint32_t device, 1453 type_t type) 1454 : ThreadBase(audioFlinger, id, device, type), 1455 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1456 // Assumes constructor is called by AudioFlinger with it's mLock held, 1457 // but it would be safer to explicitly pass initial masterMute as parameter 1458 mMasterMute(audioFlinger->masterMute_l()), 1459 // mStreamTypes[] initialized in constructor body 1460 mOutput(output), 1461 // Assumes constructor is called by AudioFlinger with it's mLock held, 1462 // but it would be safer to explicitly pass initial masterVolume as parameter 1463 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1464 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1465{ 1466 snprintf(mName, kNameLength, "AudioOut_%d", id); 1467 1468 readOutputParameters(); 1469 1470 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1471 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1472 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1473 stream = (audio_stream_type_t) (stream + 1)) { 1474 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1475 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1476 // initialized by stream_type_t default constructor 1477 // mStreamTypes[stream].valid = true; 1478 } 1479 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1480 // because mAudioFlinger doesn't have one to copy from 1481} 1482 1483AudioFlinger::PlaybackThread::~PlaybackThread() 1484{ 1485 delete [] mMixBuffer; 1486} 1487 1488status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1489{ 1490 dumpInternals(fd, args); 1491 dumpTracks(fd, args); 1492 dumpEffectChains(fd, args); 1493 return NO_ERROR; 1494} 1495 1496status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1497{ 1498 const size_t SIZE = 256; 1499 char buffer[SIZE]; 1500 String8 result; 1501 1502 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1503 result.append(buffer); 1504 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1505 for (size_t i = 0; i < mTracks.size(); ++i) { 1506 sp<Track> track = mTracks[i]; 1507 if (track != 0) { 1508 track->dump(buffer, SIZE); 1509 result.append(buffer); 1510 } 1511 } 1512 1513 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1514 result.append(buffer); 1515 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1516 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1517 sp<Track> track = mActiveTracks[i].promote(); 1518 if (track != 0) { 1519 track->dump(buffer, SIZE); 1520 result.append(buffer); 1521 } 1522 } 1523 write(fd, result.string(), result.size()); 1524 return NO_ERROR; 1525} 1526 1527status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1528{ 1529 const size_t SIZE = 256; 1530 char buffer[SIZE]; 1531 String8 result; 1532 1533 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1534 result.append(buffer); 1535 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1536 result.append(buffer); 1537 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1538 result.append(buffer); 1539 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1540 result.append(buffer); 1541 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1542 result.append(buffer); 1543 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1544 result.append(buffer); 1545 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1546 result.append(buffer); 1547 write(fd, result.string(), result.size()); 1548 1549 dumpBase(fd, args); 1550 1551 return NO_ERROR; 1552} 1553 1554// Thread virtuals 1555status_t AudioFlinger::PlaybackThread::readyToRun() 1556{ 1557 status_t status = initCheck(); 1558 if (status == NO_ERROR) { 1559 ALOGI("AudioFlinger's thread %p ready to run", this); 1560 } else { 1561 ALOGE("No working audio driver found."); 1562 } 1563 return status; 1564} 1565 1566void AudioFlinger::PlaybackThread::onFirstRef() 1567{ 1568 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1569} 1570 1571// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1572sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1573 const sp<AudioFlinger::Client>& client, 1574 audio_stream_type_t streamType, 1575 uint32_t sampleRate, 1576 audio_format_t format, 1577 uint32_t channelMask, 1578 int frameCount, 1579 const sp<IMemory>& sharedBuffer, 1580 int sessionId, 1581 bool isTimed, 1582 status_t *status) 1583{ 1584 sp<Track> track; 1585 status_t lStatus; 1586 1587 if (mType == DIRECT) { 1588 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1589 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1590 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1591 "for output %p with format %d", 1592 sampleRate, format, channelMask, mOutput, mFormat); 1593 lStatus = BAD_VALUE; 1594 goto Exit; 1595 } 1596 } 1597 } else { 1598 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1599 if (sampleRate > mSampleRate*2) { 1600 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1601 lStatus = BAD_VALUE; 1602 goto Exit; 1603 } 1604 } 1605 1606 lStatus = initCheck(); 1607 if (lStatus != NO_ERROR) { 1608 ALOGE("Audio driver not initialized."); 1609 goto Exit; 1610 } 1611 1612 { // scope for mLock 1613 Mutex::Autolock _l(mLock); 1614 1615 // all tracks in same audio session must share the same routing strategy otherwise 1616 // conflicts will happen when tracks are moved from one output to another by audio policy 1617 // manager 1618 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1619 for (size_t i = 0; i < mTracks.size(); ++i) { 1620 sp<Track> t = mTracks[i]; 1621 if (t != 0) { 1622 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1623 if (sessionId == t->sessionId() && strategy != actual) { 1624 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1625 strategy, actual); 1626 lStatus = BAD_VALUE; 1627 goto Exit; 1628 } 1629 } 1630 } 1631 1632 if (!isTimed) { 1633 track = new Track(this, client, streamType, sampleRate, format, 1634 channelMask, frameCount, sharedBuffer, sessionId); 1635 } else { 1636 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1637 channelMask, frameCount, sharedBuffer, sessionId); 1638 } 1639 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1640 lStatus = NO_MEMORY; 1641 goto Exit; 1642 } 1643 mTracks.add(track); 1644 1645 sp<EffectChain> chain = getEffectChain_l(sessionId); 1646 if (chain != 0) { 1647 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1648 track->setMainBuffer(chain->inBuffer()); 1649 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1650 chain->incTrackCnt(); 1651 } 1652 1653 // invalidate track immediately if the stream type was moved to another thread since 1654 // createTrack() was called by the client process. 1655 if (!mStreamTypes[streamType].valid) { 1656 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1657 this, streamType); 1658 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1659 } 1660 } 1661 lStatus = NO_ERROR; 1662 1663Exit: 1664 if(status) { 1665 *status = lStatus; 1666 } 1667 return track; 1668} 1669 1670uint32_t AudioFlinger::PlaybackThread::latency() const 1671{ 1672 Mutex::Autolock _l(mLock); 1673 if (initCheck() == NO_ERROR) { 1674 return mOutput->stream->get_latency(mOutput->stream); 1675 } else { 1676 return 0; 1677 } 1678} 1679 1680void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1681{ 1682 Mutex::Autolock _l(mLock); 1683 mMasterVolume = value; 1684} 1685 1686void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1687{ 1688 Mutex::Autolock _l(mLock); 1689 setMasterMute_l(muted); 1690} 1691 1692void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1693{ 1694 Mutex::Autolock _l(mLock); 1695 mStreamTypes[stream].volume = value; 1696} 1697 1698void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1699{ 1700 Mutex::Autolock _l(mLock); 1701 mStreamTypes[stream].mute = muted; 1702} 1703 1704float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1705{ 1706 Mutex::Autolock _l(mLock); 1707 return mStreamTypes[stream].volume; 1708} 1709 1710// addTrack_l() must be called with ThreadBase::mLock held 1711status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1712{ 1713 status_t status = ALREADY_EXISTS; 1714 1715 // set retry count for buffer fill 1716 track->mRetryCount = kMaxTrackStartupRetries; 1717 if (mActiveTracks.indexOf(track) < 0) { 1718 // the track is newly added, make sure it fills up all its 1719 // buffers before playing. This is to ensure the client will 1720 // effectively get the latency it requested. 1721 track->mFillingUpStatus = Track::FS_FILLING; 1722 track->mResetDone = false; 1723 mActiveTracks.add(track); 1724 if (track->mainBuffer() != mMixBuffer) { 1725 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1726 if (chain != 0) { 1727 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1728 chain->incActiveTrackCnt(); 1729 } 1730 } 1731 1732 status = NO_ERROR; 1733 } 1734 1735 ALOGV("mWaitWorkCV.broadcast"); 1736 mWaitWorkCV.broadcast(); 1737 1738 return status; 1739} 1740 1741// destroyTrack_l() must be called with ThreadBase::mLock held 1742void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1743{ 1744 track->mState = TrackBase::TERMINATED; 1745 if (mActiveTracks.indexOf(track) < 0) { 1746 removeTrack_l(track); 1747 } 1748} 1749 1750void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1751{ 1752 mTracks.remove(track); 1753 deleteTrackName_l(track->name()); 1754 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1755 if (chain != 0) { 1756 chain->decTrackCnt(); 1757 } 1758} 1759 1760String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1761{ 1762 String8 out_s8 = String8(""); 1763 char *s; 1764 1765 Mutex::Autolock _l(mLock); 1766 if (initCheck() != NO_ERROR) { 1767 return out_s8; 1768 } 1769 1770 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1771 out_s8 = String8(s); 1772 free(s); 1773 return out_s8; 1774} 1775 1776// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1777void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1778 AudioSystem::OutputDescriptor desc; 1779 void *param2 = NULL; 1780 1781 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1782 1783 switch (event) { 1784 case AudioSystem::OUTPUT_OPENED: 1785 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1786 desc.channels = mChannelMask; 1787 desc.samplingRate = mSampleRate; 1788 desc.format = mFormat; 1789 desc.frameCount = mFrameCount; 1790 desc.latency = latency(); 1791 param2 = &desc; 1792 break; 1793 1794 case AudioSystem::STREAM_CONFIG_CHANGED: 1795 param2 = ¶m; 1796 case AudioSystem::OUTPUT_CLOSED: 1797 default: 1798 break; 1799 } 1800 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1801} 1802 1803void AudioFlinger::PlaybackThread::readOutputParameters() 1804{ 1805 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1806 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1807 mChannelCount = (uint16_t)popcount(mChannelMask); 1808 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1809 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1810 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1811 1812 // FIXME - Current mixer implementation only supports stereo output: Always 1813 // Allocate a stereo buffer even if HW output is mono. 1814 delete[] mMixBuffer; 1815 mMixBuffer = new int16_t[mFrameCount * 2]; 1816 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1817 1818 // force reconfiguration of effect chains and engines to take new buffer size and audio 1819 // parameters into account 1820 // Note that mLock is not held when readOutputParameters() is called from the constructor 1821 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1822 // matter. 1823 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1824 Vector< sp<EffectChain> > effectChains = mEffectChains; 1825 for (size_t i = 0; i < effectChains.size(); i ++) { 1826 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1827 } 1828} 1829 1830status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1831{ 1832 if (halFrames == NULL || dspFrames == NULL) { 1833 return BAD_VALUE; 1834 } 1835 Mutex::Autolock _l(mLock); 1836 if (initCheck() != NO_ERROR) { 1837 return INVALID_OPERATION; 1838 } 1839 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1840 1841 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1842} 1843 1844uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1845{ 1846 Mutex::Autolock _l(mLock); 1847 uint32_t result = 0; 1848 if (getEffectChain_l(sessionId) != 0) { 1849 result = EFFECT_SESSION; 1850 } 1851 1852 for (size_t i = 0; i < mTracks.size(); ++i) { 1853 sp<Track> track = mTracks[i]; 1854 if (sessionId == track->sessionId() && 1855 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1856 result |= TRACK_SESSION; 1857 break; 1858 } 1859 } 1860 1861 return result; 1862} 1863 1864uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1865{ 1866 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1867 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1868 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1869 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1870 } 1871 for (size_t i = 0; i < mTracks.size(); i++) { 1872 sp<Track> track = mTracks[i]; 1873 if (sessionId == track->sessionId() && 1874 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1875 return AudioSystem::getStrategyForStream(track->streamType()); 1876 } 1877 } 1878 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1879} 1880 1881 1882AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1883{ 1884 Mutex::Autolock _l(mLock); 1885 return mOutput; 1886} 1887 1888AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1889{ 1890 Mutex::Autolock _l(mLock); 1891 AudioStreamOut *output = mOutput; 1892 mOutput = NULL; 1893 return output; 1894} 1895 1896// this method must always be called either with ThreadBase mLock held or inside the thread loop 1897audio_stream_t* AudioFlinger::PlaybackThread::stream() 1898{ 1899 if (mOutput == NULL) { 1900 return NULL; 1901 } 1902 return &mOutput->stream->common; 1903} 1904 1905uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1906{ 1907 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1908 // decoding and transfer time. So sleeping for half of the latency would likely cause 1909 // underruns 1910 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1911 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1912 } else { 1913 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1914 } 1915} 1916 1917// ---------------------------------------------------------------------------- 1918 1919AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1920 audio_io_handle_t id, uint32_t device, type_t type) 1921 : PlaybackThread(audioFlinger, output, id, device, type), 1922 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1923 mPrevMixerStatus(MIXER_IDLE) 1924{ 1925 // FIXME - Current mixer implementation only supports stereo output 1926 if (mChannelCount == 1) { 1927 ALOGE("Invalid audio hardware channel count"); 1928 } 1929} 1930 1931AudioFlinger::MixerThread::~MixerThread() 1932{ 1933 delete mAudioMixer; 1934} 1935 1936class CpuStats { 1937public: 1938 void sample(); 1939#ifdef DEBUG_CPU_USAGE 1940private: 1941 ThreadCpuUsage mCpu; 1942#endif 1943}; 1944 1945void CpuStats::sample() { 1946#ifdef DEBUG_CPU_USAGE 1947 const CentralTendencyStatistics& stats = mCpu.statistics(); 1948 mCpu.sampleAndEnable(); 1949 unsigned n = stats.n(); 1950 // mCpu.elapsed() is expensive, so don't call it every loop 1951 if ((n & 127) == 1) { 1952 long long elapsed = mCpu.elapsed(); 1953 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1954 double perLoop = elapsed / (double) n; 1955 double perLoop100 = perLoop * 0.01; 1956 double mean = stats.mean(); 1957 double stddev = stats.stddev(); 1958 double minimum = stats.minimum(); 1959 double maximum = stats.maximum(); 1960 mCpu.resetStatistics(); 1961 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", 1962 elapsed * .000000001, n, perLoop * .000001, 1963 mean * .001, 1964 stddev * .001, 1965 minimum * .001, 1966 maximum * .001, 1967 mean / perLoop100, 1968 stddev / perLoop100, 1969 minimum / perLoop100, 1970 maximum / perLoop100); 1971 } 1972 } 1973#endif 1974}; 1975 1976void AudioFlinger::PlaybackThread::checkSilentMode_l() 1977{ 1978 if (!mMasterMute) { 1979 char value[PROPERTY_VALUE_MAX]; 1980 if (property_get("ro.audio.silent", value, "0") > 0) { 1981 char *endptr; 1982 unsigned long ul = strtoul(value, &endptr, 0); 1983 if (*endptr == '\0' && ul != 0) { 1984 ALOGD("Silence is golden"); 1985 // The setprop command will not allow a property to be changed after 1986 // the first time it is set, so we don't have to worry about un-muting. 1987 setMasterMute_l(true); 1988 } 1989 } 1990 } 1991} 1992 1993bool AudioFlinger::MixerThread::threadLoop() 1994{ 1995 Vector< sp<Track> > tracksToRemove; 1996 nsecs_t standbyTime = systemTime(); 1997 size_t mixBufferSize = mFrameCount * mFrameSize; 1998 // FIXME: Relaxed timing because of a certain device that can't meet latency 1999 // Should be reduced to 2x after the vendor fixes the driver issue 2000 // increase threshold again due to low power audio mode. The way this warning threshold is 2001 // calculated and its usefulness should be reconsidered anyway. 2002 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2003 nsecs_t lastWarning = 0; 2004 bool longStandbyExit = false; 2005 uint32_t activeSleepTime = activeSleepTimeUs(); 2006 uint32_t idleSleepTime = idleSleepTimeUs(); 2007 uint32_t sleepTime = idleSleepTime; 2008 uint32_t sleepTimeShift = 0; 2009 Vector< sp<EffectChain> > effectChains; 2010 CpuStats cpuStats; 2011 2012 acquireWakeLock(); 2013 2014 while (!exitPending()) 2015 { 2016 cpuStats.sample(); 2017 processConfigEvents(); 2018 2019 mixer_state mixerStatus = MIXER_IDLE; 2020 { // scope for mLock 2021 2022 Mutex::Autolock _l(mLock); 2023 2024 if (checkForNewParameters_l()) { 2025 mixBufferSize = mFrameCount * mFrameSize; 2026 // FIXME: Relaxed timing because of a certain device that can't meet latency 2027 // Should be reduced to 2x after the vendor fixes the driver issue 2028 // increase threshold again due to low power audio mode. The way this warning 2029 // threshold is calculated and its usefulness should be reconsidered anyway. 2030 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2031 activeSleepTime = activeSleepTimeUs(); 2032 idleSleepTime = idleSleepTimeUs(); 2033 } 2034 2035 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2036 2037 // put audio hardware into standby after short delay 2038 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2039 mSuspended)) { 2040 if (!mStandby) { 2041 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2042 mOutput->stream->common.standby(&mOutput->stream->common); 2043 mStandby = true; 2044 mBytesWritten = 0; 2045 } 2046 2047 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2048 // we're about to wait, flush the binder command buffer 2049 IPCThreadState::self()->flushCommands(); 2050 2051 if (exitPending()) break; 2052 2053 releaseWakeLock_l(); 2054 // wait until we have something to do... 2055 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2056 mWaitWorkCV.wait(mLock); 2057 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2058 acquireWakeLock_l(); 2059 2060 mPrevMixerStatus = MIXER_IDLE; 2061 checkSilentMode_l(); 2062 2063 standbyTime = systemTime() + mStandbyTimeInNsecs; 2064 sleepTime = idleSleepTime; 2065 sleepTimeShift = 0; 2066 continue; 2067 } 2068 } 2069 2070 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2071 2072 // prevent any changes in effect chain list and in each effect chain 2073 // during mixing and effect process as the audio buffers could be deleted 2074 // or modified if an effect is created or deleted 2075 lockEffectChains_l(effectChains); 2076 } 2077 2078 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2079 // obtain the presentation timestamp of the next output buffer 2080 int64_t pts; 2081 status_t status = INVALID_OPERATION; 2082 2083 if (NULL != mOutput->stream->get_next_write_timestamp) { 2084 status = mOutput->stream->get_next_write_timestamp( 2085 mOutput->stream, &pts); 2086 } 2087 2088 if (status != NO_ERROR) { 2089 pts = AudioBufferProvider::kInvalidPTS; 2090 } 2091 2092 // mix buffers... 2093 mAudioMixer->process(pts); 2094 // increase sleep time progressively when application underrun condition clears. 2095 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2096 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2097 // such that we would underrun the audio HAL. 2098 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2099 sleepTimeShift--; 2100 } 2101 sleepTime = 0; 2102 standbyTime = systemTime() + mStandbyTimeInNsecs; 2103 //TODO: delay standby when effects have a tail 2104 } else { 2105 // If no tracks are ready, sleep once for the duration of an output 2106 // buffer size, then write 0s to the output 2107 if (sleepTime == 0) { 2108 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2109 sleepTime = activeSleepTime >> sleepTimeShift; 2110 if (sleepTime < kMinThreadSleepTimeUs) { 2111 sleepTime = kMinThreadSleepTimeUs; 2112 } 2113 // reduce sleep time in case of consecutive application underruns to avoid 2114 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2115 // duration we would end up writing less data than needed by the audio HAL if 2116 // the condition persists. 2117 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2118 sleepTimeShift++; 2119 } 2120 } else { 2121 sleepTime = idleSleepTime; 2122 } 2123 } else if (mBytesWritten != 0 || 2124 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2125 memset (mMixBuffer, 0, mixBufferSize); 2126 sleepTime = 0; 2127 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2128 } 2129 // TODO add standby time extension fct of effect tail 2130 } 2131 2132 if (mSuspended) { 2133 sleepTime = suspendSleepTimeUs(); 2134 } 2135 // sleepTime == 0 means we must write to audio hardware 2136 if (sleepTime == 0) { 2137 for (size_t i = 0; i < effectChains.size(); i ++) { 2138 effectChains[i]->process_l(); 2139 } 2140 // enable changes in effect chain 2141 unlockEffectChains(effectChains); 2142 mLastWriteTime = systemTime(); 2143 mInWrite = true; 2144 mBytesWritten += mixBufferSize; 2145 2146 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2147 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2148 mNumWrites++; 2149 mInWrite = false; 2150 nsecs_t now = systemTime(); 2151 nsecs_t delta = now - mLastWriteTime; 2152 if (!mStandby && delta > maxPeriod) { 2153 mNumDelayedWrites++; 2154 if ((now - lastWarning) > kWarningThrottleNs) { 2155 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2156 ns2ms(delta), mNumDelayedWrites, this); 2157 lastWarning = now; 2158 } 2159 if (mStandby) { 2160 longStandbyExit = true; 2161 } 2162 } 2163 mStandby = false; 2164 } else { 2165 // enable changes in effect chain 2166 unlockEffectChains(effectChains); 2167 usleep(sleepTime); 2168 } 2169 2170 // finally let go of all our tracks, without the lock held 2171 // since we can't guarantee the destructors won't acquire that 2172 // same lock. 2173 tracksToRemove.clear(); 2174 2175 // Effect chains will be actually deleted here if they were removed from 2176 // mEffectChains list during mixing or effects processing 2177 effectChains.clear(); 2178 } 2179 2180 if (!mStandby) { 2181 mOutput->stream->common.standby(&mOutput->stream->common); 2182 } 2183 2184 releaseWakeLock(); 2185 2186 ALOGV("Thread %p type %d exiting", this, mType); 2187 return false; 2188} 2189 2190// prepareTracks_l() must be called with ThreadBase::mLock held 2191AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2192 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2193{ 2194 2195 mixer_state mixerStatus = MIXER_IDLE; 2196 // find out which tracks need to be processed 2197 size_t count = activeTracks.size(); 2198 size_t mixedTracks = 0; 2199 size_t tracksWithEffect = 0; 2200 2201 float masterVolume = mMasterVolume; 2202 bool masterMute = mMasterMute; 2203 2204 if (masterMute) { 2205 masterVolume = 0; 2206 } 2207 // Delegate master volume control to effect in output mix effect chain if needed 2208 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2209 if (chain != 0) { 2210 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2211 chain->setVolume_l(&v, &v); 2212 masterVolume = (float)((v + (1 << 23)) >> 24); 2213 chain.clear(); 2214 } 2215 2216 for (size_t i=0 ; i<count ; i++) { 2217 sp<Track> t = activeTracks[i].promote(); 2218 if (t == 0) continue; 2219 2220 // this const just means the local variable doesn't change 2221 Track* const track = t.get(); 2222 audio_track_cblk_t* cblk = track->cblk(); 2223 2224 // The first time a track is added we wait 2225 // for all its buffers to be filled before processing it 2226 int name = track->name(); 2227 // make sure that we have enough frames to mix one full buffer. 2228 // enforce this condition only once to enable draining the buffer in case the client 2229 // app does not call stop() and relies on underrun to stop: 2230 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2231 // during last round 2232 uint32_t minFrames = 1; 2233 if (!track->isStopped() && !track->isPausing() && 2234 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2235 if (t->sampleRate() == (int)mSampleRate) { 2236 minFrames = mFrameCount; 2237 } else { 2238 // +1 for rounding and +1 for additional sample needed for interpolation 2239 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2240 // add frames already consumed but not yet released by the resampler 2241 // because cblk->framesReady() will include these frames 2242 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2243 // the minimum track buffer size is normally twice the number of frames necessary 2244 // to fill one buffer and the resampler should not leave more than one buffer worth 2245 // of unreleased frames after each pass, but just in case... 2246 ALOG_ASSERT(minFrames <= cblk->frameCount); 2247 } 2248 } 2249 if ((track->framesReady() >= minFrames) && track->isReady() && 2250 !track->isPaused() && !track->isTerminated()) 2251 { 2252 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2253 2254 mixedTracks++; 2255 2256 // track->mainBuffer() != mMixBuffer means there is an effect chain 2257 // connected to the track 2258 chain.clear(); 2259 if (track->mainBuffer() != mMixBuffer) { 2260 chain = getEffectChain_l(track->sessionId()); 2261 // Delegate volume control to effect in track effect chain if needed 2262 if (chain != 0) { 2263 tracksWithEffect++; 2264 } else { 2265 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2266 name, track->sessionId()); 2267 } 2268 } 2269 2270 2271 int param = AudioMixer::VOLUME; 2272 if (track->mFillingUpStatus == Track::FS_FILLED) { 2273 // no ramp for the first volume setting 2274 track->mFillingUpStatus = Track::FS_ACTIVE; 2275 if (track->mState == TrackBase::RESUMING) { 2276 track->mState = TrackBase::ACTIVE; 2277 param = AudioMixer::RAMP_VOLUME; 2278 } 2279 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2280 } else if (cblk->server != 0) { 2281 // If the track is stopped before the first frame was mixed, 2282 // do not apply ramp 2283 param = AudioMixer::RAMP_VOLUME; 2284 } 2285 2286 // compute volume for this track 2287 uint32_t vl, vr, va; 2288 if (track->isMuted() || track->isPausing() || 2289 mStreamTypes[track->streamType()].mute) { 2290 vl = vr = va = 0; 2291 if (track->isPausing()) { 2292 track->setPaused(); 2293 } 2294 } else { 2295 2296 // read original volumes with volume control 2297 float typeVolume = mStreamTypes[track->streamType()].volume; 2298 float v = masterVolume * typeVolume; 2299 uint32_t vlr = cblk->getVolumeLR(); 2300 vl = vlr & 0xFFFF; 2301 vr = vlr >> 16; 2302 // track volumes come from shared memory, so can't be trusted and must be clamped 2303 if (vl > MAX_GAIN_INT) { 2304 ALOGV("Track left volume out of range: %04X", vl); 2305 vl = MAX_GAIN_INT; 2306 } 2307 if (vr > MAX_GAIN_INT) { 2308 ALOGV("Track right volume out of range: %04X", vr); 2309 vr = MAX_GAIN_INT; 2310 } 2311 // now apply the master volume and stream type volume 2312 vl = (uint32_t)(v * vl) << 12; 2313 vr = (uint32_t)(v * vr) << 12; 2314 // assuming master volume and stream type volume each go up to 1.0, 2315 // vl and vr are now in 8.24 format 2316 2317 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2318 // send level comes from shared memory and so may be corrupt 2319 if (sendLevel > MAX_GAIN_INT) { 2320 ALOGV("Track send level out of range: %04X", sendLevel); 2321 sendLevel = MAX_GAIN_INT; 2322 } 2323 va = (uint32_t)(v * sendLevel); 2324 } 2325 // Delegate volume control to effect in track effect chain if needed 2326 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2327 // Do not ramp volume if volume is controlled by effect 2328 param = AudioMixer::VOLUME; 2329 track->mHasVolumeController = true; 2330 } else { 2331 // force no volume ramp when volume controller was just disabled or removed 2332 // from effect chain to avoid volume spike 2333 if (track->mHasVolumeController) { 2334 param = AudioMixer::VOLUME; 2335 } 2336 track->mHasVolumeController = false; 2337 } 2338 2339 // Convert volumes from 8.24 to 4.12 format 2340 // This additional clamping is needed in case chain->setVolume_l() overshot 2341 vl = (vl + (1 << 11)) >> 12; 2342 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2343 vr = (vr + (1 << 11)) >> 12; 2344 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2345 2346 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2347 2348 // XXX: these things DON'T need to be done each time 2349 mAudioMixer->setBufferProvider(name, track); 2350 mAudioMixer->enable(name); 2351 2352 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2353 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2354 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2355 mAudioMixer->setParameter( 2356 name, 2357 AudioMixer::TRACK, 2358 AudioMixer::FORMAT, (void *)track->format()); 2359 mAudioMixer->setParameter( 2360 name, 2361 AudioMixer::TRACK, 2362 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2363 mAudioMixer->setParameter( 2364 name, 2365 AudioMixer::RESAMPLE, 2366 AudioMixer::SAMPLE_RATE, 2367 (void *)(cblk->sampleRate)); 2368 mAudioMixer->setParameter( 2369 name, 2370 AudioMixer::TRACK, 2371 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2372 mAudioMixer->setParameter( 2373 name, 2374 AudioMixer::TRACK, 2375 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2376 2377 // reset retry count 2378 track->mRetryCount = kMaxTrackRetries; 2379 // If one track is ready, set the mixer ready if: 2380 // - the mixer was not ready during previous round OR 2381 // - no other track is not ready 2382 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2383 mixerStatus != MIXER_TRACKS_ENABLED) { 2384 mixerStatus = MIXER_TRACKS_READY; 2385 } 2386 } else { 2387 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2388 if (track->isStopped()) { 2389 track->reset(); 2390 } 2391 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2392 // We have consumed all the buffers of this track. 2393 // Remove it from the list of active tracks. 2394 tracksToRemove->add(track); 2395 } else { 2396 // No buffers for this track. Give it a few chances to 2397 // fill a buffer, then remove it from active list. 2398 if (--(track->mRetryCount) <= 0) { 2399 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2400 tracksToRemove->add(track); 2401 // indicate to client process that the track was disabled because of underrun 2402 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2403 // If one track is not ready, mark the mixer also not ready if: 2404 // - the mixer was ready during previous round OR 2405 // - no other track is ready 2406 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2407 mixerStatus != MIXER_TRACKS_READY) { 2408 mixerStatus = MIXER_TRACKS_ENABLED; 2409 } 2410 } 2411 mAudioMixer->disable(name); 2412 } 2413 } 2414 2415 // remove all the tracks that need to be... 2416 count = tracksToRemove->size(); 2417 if (CC_UNLIKELY(count)) { 2418 for (size_t i=0 ; i<count ; i++) { 2419 const sp<Track>& track = tracksToRemove->itemAt(i); 2420 mActiveTracks.remove(track); 2421 if (track->mainBuffer() != mMixBuffer) { 2422 chain = getEffectChain_l(track->sessionId()); 2423 if (chain != 0) { 2424 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2425 chain->decActiveTrackCnt(); 2426 } 2427 } 2428 if (track->isTerminated()) { 2429 removeTrack_l(track); 2430 } 2431 } 2432 } 2433 2434 // mix buffer must be cleared if all tracks are connected to an 2435 // effect chain as in this case the mixer will not write to 2436 // mix buffer and track effects will accumulate into it 2437 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2438 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2439 } 2440 2441 mPrevMixerStatus = mixerStatus; 2442 return mixerStatus; 2443} 2444 2445void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2446{ 2447 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2448 this, streamType, mTracks.size()); 2449 Mutex::Autolock _l(mLock); 2450 2451 size_t size = mTracks.size(); 2452 for (size_t i = 0; i < size; i++) { 2453 sp<Track> t = mTracks[i]; 2454 if (t->streamType() == streamType) { 2455 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2456 t->mCblk->cv.signal(); 2457 } 2458 } 2459} 2460 2461void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2462{ 2463 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2464 this, streamType, valid); 2465 Mutex::Autolock _l(mLock); 2466 2467 mStreamTypes[streamType].valid = valid; 2468} 2469 2470// getTrackName_l() must be called with ThreadBase::mLock held 2471int AudioFlinger::MixerThread::getTrackName_l() 2472{ 2473 return mAudioMixer->getTrackName(); 2474} 2475 2476// deleteTrackName_l() must be called with ThreadBase::mLock held 2477void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2478{ 2479 ALOGV("remove track (%d) and delete from mixer", name); 2480 mAudioMixer->deleteTrackName(name); 2481} 2482 2483// checkForNewParameters_l() must be called with ThreadBase::mLock held 2484bool AudioFlinger::MixerThread::checkForNewParameters_l() 2485{ 2486 bool reconfig = false; 2487 2488 while (!mNewParameters.isEmpty()) { 2489 status_t status = NO_ERROR; 2490 String8 keyValuePair = mNewParameters[0]; 2491 AudioParameter param = AudioParameter(keyValuePair); 2492 int value; 2493 2494 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2495 reconfig = true; 2496 } 2497 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2498 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2499 status = BAD_VALUE; 2500 } else { 2501 reconfig = true; 2502 } 2503 } 2504 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2505 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2506 status = BAD_VALUE; 2507 } else { 2508 reconfig = true; 2509 } 2510 } 2511 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2512 // do not accept frame count changes if tracks are open as the track buffer 2513 // size depends on frame count and correct behavior would not be guaranteed 2514 // if frame count is changed after track creation 2515 if (!mTracks.isEmpty()) { 2516 status = INVALID_OPERATION; 2517 } else { 2518 reconfig = true; 2519 } 2520 } 2521 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2522 // when changing the audio output device, call addBatteryData to notify 2523 // the change 2524 if ((int)mDevice != value) { 2525 uint32_t params = 0; 2526 // check whether speaker is on 2527 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2528 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2529 } 2530 2531 int deviceWithoutSpeaker 2532 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2533 // check if any other device (except speaker) is on 2534 if (value & deviceWithoutSpeaker ) { 2535 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2536 } 2537 2538 if (params != 0) { 2539 addBatteryData(params); 2540 } 2541 } 2542 2543 // forward device change to effects that have requested to be 2544 // aware of attached audio device. 2545 mDevice = (uint32_t)value; 2546 for (size_t i = 0; i < mEffectChains.size(); i++) { 2547 mEffectChains[i]->setDevice_l(mDevice); 2548 } 2549 } 2550 2551 if (status == NO_ERROR) { 2552 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2553 keyValuePair.string()); 2554 if (!mStandby && status == INVALID_OPERATION) { 2555 mOutput->stream->common.standby(&mOutput->stream->common); 2556 mStandby = true; 2557 mBytesWritten = 0; 2558 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2559 keyValuePair.string()); 2560 } 2561 if (status == NO_ERROR && reconfig) { 2562 delete mAudioMixer; 2563 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2564 mAudioMixer = NULL; 2565 readOutputParameters(); 2566 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2567 for (size_t i = 0; i < mTracks.size() ; i++) { 2568 int name = getTrackName_l(); 2569 if (name < 0) break; 2570 mTracks[i]->mName = name; 2571 // limit track sample rate to 2 x new output sample rate 2572 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2573 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2574 } 2575 } 2576 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2577 } 2578 } 2579 2580 mNewParameters.removeAt(0); 2581 2582 mParamStatus = status; 2583 mParamCond.signal(); 2584 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2585 // already timed out waiting for the status and will never signal the condition. 2586 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2587 } 2588 return reconfig; 2589} 2590 2591status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2592{ 2593 const size_t SIZE = 256; 2594 char buffer[SIZE]; 2595 String8 result; 2596 2597 PlaybackThread::dumpInternals(fd, args); 2598 2599 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2600 result.append(buffer); 2601 write(fd, result.string(), result.size()); 2602 return NO_ERROR; 2603} 2604 2605uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2606{ 2607 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2608} 2609 2610uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2611{ 2612 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2613} 2614 2615// ---------------------------------------------------------------------------- 2616AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2617 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2618 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2619 // mLeftVolFloat, mRightVolFloat 2620 // mLeftVolShort, mRightVolShort 2621{ 2622} 2623 2624AudioFlinger::DirectOutputThread::~DirectOutputThread() 2625{ 2626} 2627 2628void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2629{ 2630 // Do not apply volume on compressed audio 2631 if (!audio_is_linear_pcm(mFormat)) { 2632 return; 2633 } 2634 2635 // convert to signed 16 bit before volume calculation 2636 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2637 size_t count = mFrameCount * mChannelCount; 2638 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2639 int16_t *dst = mMixBuffer + count-1; 2640 while(count--) { 2641 *dst-- = (int16_t)(*src--^0x80) << 8; 2642 } 2643 } 2644 2645 size_t frameCount = mFrameCount; 2646 int16_t *out = mMixBuffer; 2647 if (ramp) { 2648 if (mChannelCount == 1) { 2649 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2650 int32_t vlInc = d / (int32_t)frameCount; 2651 int32_t vl = ((int32_t)mLeftVolShort << 16); 2652 do { 2653 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2654 out++; 2655 vl += vlInc; 2656 } while (--frameCount); 2657 2658 } else { 2659 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2660 int32_t vlInc = d / (int32_t)frameCount; 2661 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2662 int32_t vrInc = d / (int32_t)frameCount; 2663 int32_t vl = ((int32_t)mLeftVolShort << 16); 2664 int32_t vr = ((int32_t)mRightVolShort << 16); 2665 do { 2666 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2667 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2668 out += 2; 2669 vl += vlInc; 2670 vr += vrInc; 2671 } while (--frameCount); 2672 } 2673 } else { 2674 if (mChannelCount == 1) { 2675 do { 2676 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2677 out++; 2678 } while (--frameCount); 2679 } else { 2680 do { 2681 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2682 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2683 out += 2; 2684 } while (--frameCount); 2685 } 2686 } 2687 2688 // convert back to unsigned 8 bit after volume calculation 2689 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2690 size_t count = mFrameCount * mChannelCount; 2691 int16_t *src = mMixBuffer; 2692 uint8_t *dst = (uint8_t *)mMixBuffer; 2693 while(count--) { 2694 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2695 } 2696 } 2697 2698 mLeftVolShort = leftVol; 2699 mRightVolShort = rightVol; 2700} 2701 2702bool AudioFlinger::DirectOutputThread::threadLoop() 2703{ 2704 sp<Track> trackToRemove; 2705 sp<Track> activeTrack; 2706 nsecs_t standbyTime = systemTime(); 2707 size_t mixBufferSize = mFrameCount*mFrameSize; 2708 uint32_t activeSleepTime = activeSleepTimeUs(); 2709 uint32_t idleSleepTime = idleSleepTimeUs(); 2710 uint32_t sleepTime = idleSleepTime; 2711 // use shorter standby delay as on normal output to release 2712 // hardware resources as soon as possible 2713 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2714 2715 acquireWakeLock(); 2716 2717 while (!exitPending()) 2718 { 2719 bool rampVolume; 2720 uint16_t leftVol; 2721 uint16_t rightVol; 2722 Vector< sp<EffectChain> > effectChains; 2723 2724 processConfigEvents(); 2725 2726 mixer_state mixerStatus = MIXER_IDLE; 2727 { // scope for the mLock 2728 2729 Mutex::Autolock _l(mLock); 2730 2731 if (checkForNewParameters_l()) { 2732 mixBufferSize = mFrameCount*mFrameSize; 2733 activeSleepTime = activeSleepTimeUs(); 2734 idleSleepTime = idleSleepTimeUs(); 2735 standbyDelay = microseconds(activeSleepTime*2); 2736 } 2737 2738 // put audio hardware into standby after short delay 2739 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2740 mSuspended)) { 2741 // wait until we have something to do... 2742 if (!mStandby) { 2743 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2744 mOutput->stream->common.standby(&mOutput->stream->common); 2745 mStandby = true; 2746 mBytesWritten = 0; 2747 } 2748 2749 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2750 // we're about to wait, flush the binder command buffer 2751 IPCThreadState::self()->flushCommands(); 2752 2753 if (exitPending()) break; 2754 2755 releaseWakeLock_l(); 2756 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2757 mWaitWorkCV.wait(mLock); 2758 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2759 acquireWakeLock_l(); 2760 2761 checkSilentMode_l(); 2762 2763 standbyTime = systemTime() + standbyDelay; 2764 sleepTime = idleSleepTime; 2765 continue; 2766 } 2767 } 2768 2769 effectChains = mEffectChains; 2770 2771 // find out which tracks need to be processed 2772 if (mActiveTracks.size() != 0) { 2773 sp<Track> t = mActiveTracks[0].promote(); 2774 if (t == 0) continue; 2775 2776 Track* const track = t.get(); 2777 audio_track_cblk_t* cblk = track->cblk(); 2778 2779 // The first time a track is added we wait 2780 // for all its buffers to be filled before processing it 2781 if (cblk->framesReady() && track->isReady() && 2782 !track->isPaused() && !track->isTerminated()) 2783 { 2784 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2785 2786 if (track->mFillingUpStatus == Track::FS_FILLED) { 2787 track->mFillingUpStatus = Track::FS_ACTIVE; 2788 mLeftVolFloat = mRightVolFloat = 0; 2789 mLeftVolShort = mRightVolShort = 0; 2790 if (track->mState == TrackBase::RESUMING) { 2791 track->mState = TrackBase::ACTIVE; 2792 rampVolume = true; 2793 } 2794 } else if (cblk->server != 0) { 2795 // If the track is stopped before the first frame was mixed, 2796 // do not apply ramp 2797 rampVolume = true; 2798 } 2799 // compute volume for this track 2800 float left, right; 2801 if (track->isMuted() || mMasterMute || track->isPausing() || 2802 mStreamTypes[track->streamType()].mute) { 2803 left = right = 0; 2804 if (track->isPausing()) { 2805 track->setPaused(); 2806 } 2807 } else { 2808 float typeVolume = mStreamTypes[track->streamType()].volume; 2809 float v = mMasterVolume * typeVolume; 2810 uint32_t vlr = cblk->getVolumeLR(); 2811 float v_clamped = v * (vlr & 0xFFFF); 2812 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2813 left = v_clamped/MAX_GAIN; 2814 v_clamped = v * (vlr >> 16); 2815 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2816 right = v_clamped/MAX_GAIN; 2817 } 2818 2819 if (left != mLeftVolFloat || right != mRightVolFloat) { 2820 mLeftVolFloat = left; 2821 mRightVolFloat = right; 2822 2823 // If audio HAL implements volume control, 2824 // force software volume to nominal value 2825 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2826 left = 1.0f; 2827 right = 1.0f; 2828 } 2829 2830 // Convert volumes from float to 8.24 2831 uint32_t vl = (uint32_t)(left * (1 << 24)); 2832 uint32_t vr = (uint32_t)(right * (1 << 24)); 2833 2834 // Delegate volume control to effect in track effect chain if needed 2835 // only one effect chain can be present on DirectOutputThread, so if 2836 // there is one, the track is connected to it 2837 if (!effectChains.isEmpty()) { 2838 // Do not ramp volume if volume is controlled by effect 2839 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2840 rampVolume = false; 2841 } 2842 } 2843 2844 // Convert volumes from 8.24 to 4.12 format 2845 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2846 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2847 leftVol = (uint16_t)v_clamped; 2848 v_clamped = (vr + (1 << 11)) >> 12; 2849 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2850 rightVol = (uint16_t)v_clamped; 2851 } else { 2852 leftVol = mLeftVolShort; 2853 rightVol = mRightVolShort; 2854 rampVolume = false; 2855 } 2856 2857 // reset retry count 2858 track->mRetryCount = kMaxTrackRetriesDirect; 2859 activeTrack = t; 2860 mixerStatus = MIXER_TRACKS_READY; 2861 } else { 2862 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2863 if (track->isStopped()) { 2864 track->reset(); 2865 } 2866 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2867 // We have consumed all the buffers of this track. 2868 // Remove it from the list of active tracks. 2869 trackToRemove = track; 2870 } else { 2871 // No buffers for this track. Give it a few chances to 2872 // fill a buffer, then remove it from active list. 2873 if (--(track->mRetryCount) <= 0) { 2874 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2875 trackToRemove = track; 2876 } else { 2877 mixerStatus = MIXER_TRACKS_ENABLED; 2878 } 2879 } 2880 } 2881 } 2882 2883 // remove all the tracks that need to be... 2884 if (CC_UNLIKELY(trackToRemove != 0)) { 2885 mActiveTracks.remove(trackToRemove); 2886 if (!effectChains.isEmpty()) { 2887 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2888 trackToRemove->sessionId()); 2889 effectChains[0]->decActiveTrackCnt(); 2890 } 2891 if (trackToRemove->isTerminated()) { 2892 removeTrack_l(trackToRemove); 2893 } 2894 } 2895 2896 lockEffectChains_l(effectChains); 2897 } 2898 2899 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2900 AudioBufferProvider::Buffer buffer; 2901 size_t frameCount = mFrameCount; 2902 int8_t *curBuf = (int8_t *)mMixBuffer; 2903 // output audio to hardware 2904 while (frameCount) { 2905 buffer.frameCount = frameCount; 2906 activeTrack->getNextBuffer(&buffer, 2907 AudioBufferProvider::kInvalidPTS); 2908 if (CC_UNLIKELY(buffer.raw == NULL)) { 2909 memset(curBuf, 0, frameCount * mFrameSize); 2910 break; 2911 } 2912 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2913 frameCount -= buffer.frameCount; 2914 curBuf += buffer.frameCount * mFrameSize; 2915 activeTrack->releaseBuffer(&buffer); 2916 } 2917 sleepTime = 0; 2918 standbyTime = systemTime() + standbyDelay; 2919 } else { 2920 if (sleepTime == 0) { 2921 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2922 sleepTime = activeSleepTime; 2923 } else { 2924 sleepTime = idleSleepTime; 2925 } 2926 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2927 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2928 sleepTime = 0; 2929 } 2930 } 2931 2932 if (mSuspended) { 2933 sleepTime = suspendSleepTimeUs(); 2934 } 2935 // sleepTime == 0 means we must write to audio hardware 2936 if (sleepTime == 0) { 2937 if (mixerStatus == MIXER_TRACKS_READY) { 2938 applyVolume(leftVol, rightVol, rampVolume); 2939 } 2940 for (size_t i = 0; i < effectChains.size(); i ++) { 2941 effectChains[i]->process_l(); 2942 } 2943 unlockEffectChains(effectChains); 2944 2945 mLastWriteTime = systemTime(); 2946 mInWrite = true; 2947 mBytesWritten += mixBufferSize; 2948 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2949 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2950 mNumWrites++; 2951 mInWrite = false; 2952 mStandby = false; 2953 } else { 2954 unlockEffectChains(effectChains); 2955 usleep(sleepTime); 2956 } 2957 2958 // finally let go of removed track, without the lock held 2959 // since we can't guarantee the destructors won't acquire that 2960 // same lock. 2961 trackToRemove.clear(); 2962 activeTrack.clear(); 2963 2964 // Effect chains will be actually deleted here if they were removed from 2965 // mEffectChains list during mixing or effects processing 2966 effectChains.clear(); 2967 } 2968 2969 if (!mStandby) { 2970 mOutput->stream->common.standby(&mOutput->stream->common); 2971 } 2972 2973 releaseWakeLock(); 2974 2975 ALOGV("Thread %p type %d exiting", this, mType); 2976 return false; 2977} 2978 2979// getTrackName_l() must be called with ThreadBase::mLock held 2980int AudioFlinger::DirectOutputThread::getTrackName_l() 2981{ 2982 return 0; 2983} 2984 2985// deleteTrackName_l() must be called with ThreadBase::mLock held 2986void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2987{ 2988} 2989 2990// checkForNewParameters_l() must be called with ThreadBase::mLock held 2991bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2992{ 2993 bool reconfig = false; 2994 2995 while (!mNewParameters.isEmpty()) { 2996 status_t status = NO_ERROR; 2997 String8 keyValuePair = mNewParameters[0]; 2998 AudioParameter param = AudioParameter(keyValuePair); 2999 int value; 3000 3001 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3002 // do not accept frame count changes if tracks are open as the track buffer 3003 // size depends on frame count and correct behavior would not be garantied 3004 // if frame count is changed after track creation 3005 if (!mTracks.isEmpty()) { 3006 status = INVALID_OPERATION; 3007 } else { 3008 reconfig = true; 3009 } 3010 } 3011 if (status == NO_ERROR) { 3012 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3013 keyValuePair.string()); 3014 if (!mStandby && status == INVALID_OPERATION) { 3015 mOutput->stream->common.standby(&mOutput->stream->common); 3016 mStandby = true; 3017 mBytesWritten = 0; 3018 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3019 keyValuePair.string()); 3020 } 3021 if (status == NO_ERROR && reconfig) { 3022 readOutputParameters(); 3023 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3024 } 3025 } 3026 3027 mNewParameters.removeAt(0); 3028 3029 mParamStatus = status; 3030 mParamCond.signal(); 3031 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3032 // already timed out waiting for the status and will never signal the condition. 3033 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3034 } 3035 return reconfig; 3036} 3037 3038uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3039{ 3040 uint32_t time; 3041 if (audio_is_linear_pcm(mFormat)) { 3042 time = PlaybackThread::activeSleepTimeUs(); 3043 } else { 3044 time = 10000; 3045 } 3046 return time; 3047} 3048 3049uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3050{ 3051 uint32_t time; 3052 if (audio_is_linear_pcm(mFormat)) { 3053 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3054 } else { 3055 time = 10000; 3056 } 3057 return time; 3058} 3059 3060uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3061{ 3062 uint32_t time; 3063 if (audio_is_linear_pcm(mFormat)) { 3064 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3065 } else { 3066 time = 10000; 3067 } 3068 return time; 3069} 3070 3071 3072// ---------------------------------------------------------------------------- 3073 3074AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3075 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3076 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3077 mWaitTimeMs(UINT_MAX) 3078{ 3079 addOutputTrack(mainThread); 3080} 3081 3082AudioFlinger::DuplicatingThread::~DuplicatingThread() 3083{ 3084 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3085 mOutputTracks[i]->destroy(); 3086 } 3087} 3088 3089bool AudioFlinger::DuplicatingThread::threadLoop() 3090{ 3091 Vector< sp<Track> > tracksToRemove; 3092 nsecs_t standbyTime = systemTime(); 3093 size_t mixBufferSize = mFrameCount*mFrameSize; 3094 SortedVector< sp<OutputTrack> > outputTracks; 3095 uint32_t writeFrames = 0; 3096 uint32_t activeSleepTime = activeSleepTimeUs(); 3097 uint32_t idleSleepTime = idleSleepTimeUs(); 3098 uint32_t sleepTime = idleSleepTime; 3099 Vector< sp<EffectChain> > effectChains; 3100 3101 acquireWakeLock(); 3102 3103 while (!exitPending()) 3104 { 3105 processConfigEvents(); 3106 3107 mixer_state mixerStatus = MIXER_IDLE; 3108 { // scope for the mLock 3109 3110 Mutex::Autolock _l(mLock); 3111 3112 if (checkForNewParameters_l()) { 3113 mixBufferSize = mFrameCount*mFrameSize; 3114 updateWaitTime(); 3115 activeSleepTime = activeSleepTimeUs(); 3116 idleSleepTime = idleSleepTimeUs(); 3117 } 3118 3119 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3120 3121 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3122 outputTracks.add(mOutputTracks[i]); 3123 } 3124 3125 // put audio hardware into standby after short delay 3126 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3127 mSuspended)) { 3128 if (!mStandby) { 3129 for (size_t i = 0; i < outputTracks.size(); i++) { 3130 outputTracks[i]->stop(); 3131 } 3132 mStandby = true; 3133 mBytesWritten = 0; 3134 } 3135 3136 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3137 // we're about to wait, flush the binder command buffer 3138 IPCThreadState::self()->flushCommands(); 3139 outputTracks.clear(); 3140 3141 if (exitPending()) break; 3142 3143 releaseWakeLock_l(); 3144 // wait until we have something to do... 3145 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 3146 mWaitWorkCV.wait(mLock); 3147 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 3148 acquireWakeLock_l(); 3149 3150 checkSilentMode_l(); 3151 3152 standbyTime = systemTime() + mStandbyTimeInNsecs; 3153 sleepTime = idleSleepTime; 3154 continue; 3155 } 3156 } 3157 3158 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3159 3160 // prevent any changes in effect chain list and in each effect chain 3161 // during mixing and effect process as the audio buffers could be deleted 3162 // or modified if an effect is created or deleted 3163 lockEffectChains_l(effectChains); 3164 } 3165 3166 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3167 // mix buffers... 3168 if (outputsReady(outputTracks)) { 3169 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3170 } else { 3171 memset(mMixBuffer, 0, mixBufferSize); 3172 } 3173 sleepTime = 0; 3174 writeFrames = mFrameCount; 3175 } else { 3176 if (sleepTime == 0) { 3177 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3178 sleepTime = activeSleepTime; 3179 } else { 3180 sleepTime = idleSleepTime; 3181 } 3182 } else if (mBytesWritten != 0) { 3183 // flush remaining overflow buffers in output tracks 3184 for (size_t i = 0; i < outputTracks.size(); i++) { 3185 if (outputTracks[i]->isActive()) { 3186 sleepTime = 0; 3187 writeFrames = 0; 3188 memset(mMixBuffer, 0, mixBufferSize); 3189 break; 3190 } 3191 } 3192 } 3193 } 3194 3195 if (mSuspended) { 3196 sleepTime = suspendSleepTimeUs(); 3197 } 3198 // sleepTime == 0 means we must write to audio hardware 3199 if (sleepTime == 0) { 3200 for (size_t i = 0; i < effectChains.size(); i ++) { 3201 effectChains[i]->process_l(); 3202 } 3203 // enable changes in effect chain 3204 unlockEffectChains(effectChains); 3205 3206 standbyTime = systemTime() + mStandbyTimeInNsecs; 3207 for (size_t i = 0; i < outputTracks.size(); i++) { 3208 outputTracks[i]->write(mMixBuffer, writeFrames); 3209 } 3210 mStandby = false; 3211 mBytesWritten += mixBufferSize; 3212 } else { 3213 // enable changes in effect chain 3214 unlockEffectChains(effectChains); 3215 usleep(sleepTime); 3216 } 3217 3218 // finally let go of all our tracks, without the lock held 3219 // since we can't guarantee the destructors won't acquire that 3220 // same lock. 3221 tracksToRemove.clear(); 3222 outputTracks.clear(); 3223 3224 // Effect chains will be actually deleted here if they were removed from 3225 // mEffectChains list during mixing or effects processing 3226 effectChains.clear(); 3227 } 3228 3229 releaseWakeLock(); 3230 3231 ALOGV("Thread %p type %d exiting", this, mType); 3232 return false; 3233} 3234 3235void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3236{ 3237 // FIXME explain this formula 3238 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3239 OutputTrack *outputTrack = new OutputTrack(thread, 3240 this, 3241 mSampleRate, 3242 mFormat, 3243 mChannelMask, 3244 frameCount); 3245 if (outputTrack->cblk() != NULL) { 3246 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3247 mOutputTracks.add(outputTrack); 3248 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3249 updateWaitTime(); 3250 } 3251} 3252 3253void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3254{ 3255 Mutex::Autolock _l(mLock); 3256 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3257 if (mOutputTracks[i]->thread() == thread) { 3258 mOutputTracks[i]->destroy(); 3259 mOutputTracks.removeAt(i); 3260 updateWaitTime(); 3261 return; 3262 } 3263 } 3264 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3265} 3266 3267void AudioFlinger::DuplicatingThread::updateWaitTime() 3268{ 3269 mWaitTimeMs = UINT_MAX; 3270 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3271 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3272 if (strong != 0) { 3273 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3274 if (waitTimeMs < mWaitTimeMs) { 3275 mWaitTimeMs = waitTimeMs; 3276 } 3277 } 3278 } 3279} 3280 3281 3282bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3283{ 3284 for (size_t i = 0; i < outputTracks.size(); i++) { 3285 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3286 if (thread == 0) { 3287 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3288 return false; 3289 } 3290 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3291 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3292 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3293 return false; 3294 } 3295 } 3296 return true; 3297} 3298 3299uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3300{ 3301 return (mWaitTimeMs * 1000) / 2; 3302} 3303 3304// ---------------------------------------------------------------------------- 3305 3306// TrackBase constructor must be called with AudioFlinger::mLock held 3307AudioFlinger::ThreadBase::TrackBase::TrackBase( 3308 ThreadBase *thread, 3309 const sp<Client>& client, 3310 uint32_t sampleRate, 3311 audio_format_t format, 3312 uint32_t channelMask, 3313 int frameCount, 3314 const sp<IMemory>& sharedBuffer, 3315 int sessionId) 3316 : RefBase(), 3317 mThread(thread), 3318 mClient(client), 3319 mCblk(NULL), 3320 // mBuffer 3321 // mBufferEnd 3322 mFrameCount(0), 3323 mState(IDLE), 3324 mFormat(format), 3325 mStepServerFailed(false), 3326 mSessionId(sessionId) 3327 // mChannelCount 3328 // mChannelMask 3329{ 3330 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3331 3332 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3333 size_t size = sizeof(audio_track_cblk_t); 3334 uint8_t channelCount = popcount(channelMask); 3335 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3336 if (sharedBuffer == 0) { 3337 size += bufferSize; 3338 } 3339 3340 if (client != NULL) { 3341 mCblkMemory = client->heap()->allocate(size); 3342 if (mCblkMemory != 0) { 3343 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3344 if (mCblk != NULL) { // construct the shared structure in-place. 3345 new(mCblk) audio_track_cblk_t(); 3346 // clear all buffers 3347 mCblk->frameCount = frameCount; 3348 mCblk->sampleRate = sampleRate; 3349 mChannelCount = channelCount; 3350 mChannelMask = channelMask; 3351 if (sharedBuffer == 0) { 3352 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3353 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3354 // Force underrun condition to avoid false underrun callback until first data is 3355 // written to buffer (other flags are cleared) 3356 mCblk->flags = CBLK_UNDERRUN_ON; 3357 } else { 3358 mBuffer = sharedBuffer->pointer(); 3359 } 3360 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3361 } 3362 } else { 3363 ALOGE("not enough memory for AudioTrack size=%u", size); 3364 client->heap()->dump("AudioTrack"); 3365 return; 3366 } 3367 } else { 3368 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3369 // construct the shared structure in-place. 3370 new(mCblk) audio_track_cblk_t(); 3371 // clear all buffers 3372 mCblk->frameCount = frameCount; 3373 mCblk->sampleRate = sampleRate; 3374 mChannelCount = channelCount; 3375 mChannelMask = channelMask; 3376 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3377 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3378 // Force underrun condition to avoid false underrun callback until first data is 3379 // written to buffer (other flags are cleared) 3380 mCblk->flags = CBLK_UNDERRUN_ON; 3381 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3382 } 3383} 3384 3385AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3386{ 3387 if (mCblk != NULL) { 3388 if (mClient == 0) { 3389 delete mCblk; 3390 } else { 3391 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3392 } 3393 } 3394 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3395 if (mClient != 0) { 3396 // Client destructor must run with AudioFlinger mutex locked 3397 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3398 // If the client's reference count drops to zero, the associated destructor 3399 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3400 // relying on the automatic clear() at end of scope. 3401 mClient.clear(); 3402 } 3403} 3404 3405void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3406{ 3407 buffer->raw = NULL; 3408 mFrameCount = buffer->frameCount; 3409 step(); 3410 buffer->frameCount = 0; 3411} 3412 3413bool AudioFlinger::ThreadBase::TrackBase::step() { 3414 bool result; 3415 audio_track_cblk_t* cblk = this->cblk(); 3416 3417 result = cblk->stepServer(mFrameCount); 3418 if (!result) { 3419 ALOGV("stepServer failed acquiring cblk mutex"); 3420 mStepServerFailed = true; 3421 } 3422 return result; 3423} 3424 3425void AudioFlinger::ThreadBase::TrackBase::reset() { 3426 audio_track_cblk_t* cblk = this->cblk(); 3427 3428 cblk->user = 0; 3429 cblk->server = 0; 3430 cblk->userBase = 0; 3431 cblk->serverBase = 0; 3432 mStepServerFailed = false; 3433 ALOGV("TrackBase::reset"); 3434} 3435 3436int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3437 return (int)mCblk->sampleRate; 3438} 3439 3440void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3441 audio_track_cblk_t* cblk = this->cblk(); 3442 size_t frameSize = cblk->frameSize; 3443 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3444 int8_t *bufferEnd = bufferStart + frames * frameSize; 3445 3446 // Check validity of returned pointer in case the track control block would have been corrupted. 3447 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3448 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3449 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3450 server %d, serverBase %d, user %d, userBase %d", 3451 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3452 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3453 return NULL; 3454 } 3455 3456 return bufferStart; 3457} 3458 3459// ---------------------------------------------------------------------------- 3460 3461// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3462AudioFlinger::PlaybackThread::Track::Track( 3463 PlaybackThread *thread, 3464 const sp<Client>& client, 3465 audio_stream_type_t streamType, 3466 uint32_t sampleRate, 3467 audio_format_t format, 3468 uint32_t channelMask, 3469 int frameCount, 3470 const sp<IMemory>& sharedBuffer, 3471 int sessionId) 3472 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3473 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3474 mAuxEffectId(0), mHasVolumeController(false) 3475{ 3476 if (mCblk != NULL) { 3477 if (thread != NULL) { 3478 mName = thread->getTrackName_l(); 3479 mMainBuffer = thread->mixBuffer(); 3480 } 3481 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3482 if (mName < 0) { 3483 ALOGE("no more track names available"); 3484 } 3485 mStreamType = streamType; 3486 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3487 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3488 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3489 } 3490} 3491 3492AudioFlinger::PlaybackThread::Track::~Track() 3493{ 3494 ALOGV("PlaybackThread::Track destructor"); 3495 sp<ThreadBase> thread = mThread.promote(); 3496 if (thread != 0) { 3497 Mutex::Autolock _l(thread->mLock); 3498 mState = TERMINATED; 3499 } 3500} 3501 3502void AudioFlinger::PlaybackThread::Track::destroy() 3503{ 3504 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3505 // by removing it from mTracks vector, so there is a risk that this Tracks's 3506 // destructor is called. As the destructor needs to lock mLock, 3507 // we must acquire a strong reference on this Track before locking mLock 3508 // here so that the destructor is called only when exiting this function. 3509 // On the other hand, as long as Track::destroy() is only called by 3510 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3511 // this Track with its member mTrack. 3512 sp<Track> keep(this); 3513 { // scope for mLock 3514 sp<ThreadBase> thread = mThread.promote(); 3515 if (thread != 0) { 3516 if (!isOutputTrack()) { 3517 if (mState == ACTIVE || mState == RESUMING) { 3518 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3519 3520 // to track the speaker usage 3521 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3522 } 3523 AudioSystem::releaseOutput(thread->id()); 3524 } 3525 Mutex::Autolock _l(thread->mLock); 3526 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3527 playbackThread->destroyTrack_l(this); 3528 } 3529 } 3530} 3531 3532void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3533{ 3534 uint32_t vlr = mCblk->getVolumeLR(); 3535 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", 3536 mName - AudioMixer::TRACK0, 3537 (mClient == 0) ? getpid_cached : mClient->pid(), 3538 mStreamType, 3539 mFormat, 3540 mChannelMask, 3541 mSessionId, 3542 mFrameCount, 3543 mState, 3544 mMute, 3545 mFillingUpStatus, 3546 mCblk->sampleRate, 3547 vlr & 0xFFFF, 3548 vlr >> 16, 3549 mCblk->server, 3550 mCblk->user, 3551 (int)mMainBuffer, 3552 (int)mAuxBuffer); 3553} 3554 3555status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3556 AudioBufferProvider::Buffer* buffer, int64_t pts) 3557{ 3558 audio_track_cblk_t* cblk = this->cblk(); 3559 uint32_t framesReady; 3560 uint32_t framesReq = buffer->frameCount; 3561 3562 // Check if last stepServer failed, try to step now 3563 if (mStepServerFailed) { 3564 if (!step()) goto getNextBuffer_exit; 3565 ALOGV("stepServer recovered"); 3566 mStepServerFailed = false; 3567 } 3568 3569 framesReady = cblk->framesReady(); 3570 3571 if (CC_LIKELY(framesReady)) { 3572 uint32_t s = cblk->server; 3573 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3574 3575 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3576 if (framesReq > framesReady) { 3577 framesReq = framesReady; 3578 } 3579 if (s + framesReq > bufferEnd) { 3580 framesReq = bufferEnd - s; 3581 } 3582 3583 buffer->raw = getBuffer(s, framesReq); 3584 if (buffer->raw == NULL) goto getNextBuffer_exit; 3585 3586 buffer->frameCount = framesReq; 3587 return NO_ERROR; 3588 } 3589 3590getNextBuffer_exit: 3591 buffer->raw = NULL; 3592 buffer->frameCount = 0; 3593 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3594 return NOT_ENOUGH_DATA; 3595} 3596 3597uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3598 return mCblk->framesReady(); 3599} 3600 3601bool AudioFlinger::PlaybackThread::Track::isReady() const { 3602 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3603 3604 if (framesReady() >= mCblk->frameCount || 3605 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3606 mFillingUpStatus = FS_FILLED; 3607 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3608 return true; 3609 } 3610 return false; 3611} 3612 3613status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3614{ 3615 status_t status = NO_ERROR; 3616 ALOGV("start(%d), calling pid %d session %d tid %d", 3617 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3618 sp<ThreadBase> thread = mThread.promote(); 3619 if (thread != 0) { 3620 Mutex::Autolock _l(thread->mLock); 3621 track_state state = mState; 3622 // here the track could be either new, or restarted 3623 // in both cases "unstop" the track 3624 if (mState == PAUSED) { 3625 mState = TrackBase::RESUMING; 3626 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3627 } else { 3628 mState = TrackBase::ACTIVE; 3629 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3630 } 3631 3632 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3633 thread->mLock.unlock(); 3634 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3635 thread->mLock.lock(); 3636 3637 // to track the speaker usage 3638 if (status == NO_ERROR) { 3639 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3640 } 3641 } 3642 if (status == NO_ERROR) { 3643 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3644 playbackThread->addTrack_l(this); 3645 } else { 3646 mState = state; 3647 } 3648 } else { 3649 status = BAD_VALUE; 3650 } 3651 return status; 3652} 3653 3654void AudioFlinger::PlaybackThread::Track::stop() 3655{ 3656 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3657 sp<ThreadBase> thread = mThread.promote(); 3658 if (thread != 0) { 3659 Mutex::Autolock _l(thread->mLock); 3660 track_state state = mState; 3661 if (mState > STOPPED) { 3662 mState = STOPPED; 3663 // If the track is not active (PAUSED and buffers full), flush buffers 3664 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3665 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3666 reset(); 3667 } 3668 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3669 } 3670 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3671 thread->mLock.unlock(); 3672 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3673 thread->mLock.lock(); 3674 3675 // to track the speaker usage 3676 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3677 } 3678 } 3679} 3680 3681void AudioFlinger::PlaybackThread::Track::pause() 3682{ 3683 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3684 sp<ThreadBase> thread = mThread.promote(); 3685 if (thread != 0) { 3686 Mutex::Autolock _l(thread->mLock); 3687 if (mState == ACTIVE || mState == RESUMING) { 3688 mState = PAUSING; 3689 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3690 if (!isOutputTrack()) { 3691 thread->mLock.unlock(); 3692 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3693 thread->mLock.lock(); 3694 3695 // to track the speaker usage 3696 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3697 } 3698 } 3699 } 3700} 3701 3702void AudioFlinger::PlaybackThread::Track::flush() 3703{ 3704 ALOGV("flush(%d)", mName); 3705 sp<ThreadBase> thread = mThread.promote(); 3706 if (thread != 0) { 3707 Mutex::Autolock _l(thread->mLock); 3708 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3709 return; 3710 } 3711 // No point remaining in PAUSED state after a flush => go to 3712 // STOPPED state 3713 mState = STOPPED; 3714 3715 // do not reset the track if it is still in the process of being stopped or paused. 3716 // this will be done by prepareTracks_l() when the track is stopped. 3717 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3718 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3719 reset(); 3720 } 3721 } 3722} 3723 3724void AudioFlinger::PlaybackThread::Track::reset() 3725{ 3726 // Do not reset twice to avoid discarding data written just after a flush and before 3727 // the audioflinger thread detects the track is stopped. 3728 if (!mResetDone) { 3729 TrackBase::reset(); 3730 // Force underrun condition to avoid false underrun callback until first data is 3731 // written to buffer 3732 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3733 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3734 mFillingUpStatus = FS_FILLING; 3735 mResetDone = true; 3736 } 3737} 3738 3739void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3740{ 3741 mMute = muted; 3742} 3743 3744status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3745{ 3746 status_t status = DEAD_OBJECT; 3747 sp<ThreadBase> thread = mThread.promote(); 3748 if (thread != 0) { 3749 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3750 status = playbackThread->attachAuxEffect(this, EffectId); 3751 } 3752 return status; 3753} 3754 3755void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3756{ 3757 mAuxEffectId = EffectId; 3758 mAuxBuffer = buffer; 3759} 3760 3761// timed audio tracks 3762 3763sp<AudioFlinger::PlaybackThread::TimedTrack> 3764AudioFlinger::PlaybackThread::TimedTrack::create( 3765 PlaybackThread *thread, 3766 const sp<Client>& client, 3767 audio_stream_type_t streamType, 3768 uint32_t sampleRate, 3769 audio_format_t format, 3770 uint32_t channelMask, 3771 int frameCount, 3772 const sp<IMemory>& sharedBuffer, 3773 int sessionId) { 3774 if (!client->reserveTimedTrack()) 3775 return NULL; 3776 3777 sp<TimedTrack> track = new TimedTrack( 3778 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3779 sharedBuffer, sessionId); 3780 3781 if (track == NULL) { 3782 client->releaseTimedTrack(); 3783 return NULL; 3784 } 3785 3786 return track; 3787} 3788 3789AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3790 PlaybackThread *thread, 3791 const sp<Client>& client, 3792 audio_stream_type_t streamType, 3793 uint32_t sampleRate, 3794 audio_format_t format, 3795 uint32_t channelMask, 3796 int frameCount, 3797 const sp<IMemory>& sharedBuffer, 3798 int sessionId) 3799 : Track(thread, client, streamType, sampleRate, format, channelMask, 3800 frameCount, sharedBuffer, sessionId), 3801 mTimedSilenceBuffer(NULL), 3802 mTimedSilenceBufferSize(0), 3803 mTimedAudioOutputOnTime(false), 3804 mMediaTimeTransformValid(false) 3805{ 3806 LocalClock lc; 3807 mLocalTimeFreq = lc.getLocalFreq(); 3808 3809 mLocalTimeToSampleTransform.a_zero = 0; 3810 mLocalTimeToSampleTransform.b_zero = 0; 3811 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3812 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3813 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3814 &mLocalTimeToSampleTransform.a_to_b_denom); 3815} 3816 3817AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3818 mClient->releaseTimedTrack(); 3819 delete [] mTimedSilenceBuffer; 3820} 3821 3822status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3823 size_t size, sp<IMemory>* buffer) { 3824 3825 Mutex::Autolock _l(mTimedBufferQueueLock); 3826 3827 trimTimedBufferQueue_l(); 3828 3829 // lazily initialize the shared memory heap for timed buffers 3830 if (mTimedMemoryDealer == NULL) { 3831 const int kTimedBufferHeapSize = 512 << 10; 3832 3833 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3834 "AudioFlingerTimed"); 3835 if (mTimedMemoryDealer == NULL) 3836 return NO_MEMORY; 3837 } 3838 3839 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3840 if (newBuffer == NULL) { 3841 newBuffer = mTimedMemoryDealer->allocate(size); 3842 if (newBuffer == NULL) 3843 return NO_MEMORY; 3844 } 3845 3846 *buffer = newBuffer; 3847 return NO_ERROR; 3848} 3849 3850// caller must hold mTimedBufferQueueLock 3851void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3852 int64_t mediaTimeNow; 3853 { 3854 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3855 if (!mMediaTimeTransformValid) 3856 return; 3857 3858 int64_t targetTimeNow; 3859 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3860 ? mCCHelper.getCommonTime(&targetTimeNow) 3861 : mCCHelper.getLocalTime(&targetTimeNow); 3862 3863 if (OK != res) 3864 return; 3865 3866 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3867 &mediaTimeNow)) { 3868 return; 3869 } 3870 } 3871 3872 size_t trimIndex; 3873 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3874 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3875 break; 3876 } 3877 3878 if (trimIndex) { 3879 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3880 } 3881} 3882 3883status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3884 const sp<IMemory>& buffer, int64_t pts) { 3885 3886 { 3887 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3888 if (!mMediaTimeTransformValid) 3889 return INVALID_OPERATION; 3890 } 3891 3892 Mutex::Autolock _l(mTimedBufferQueueLock); 3893 3894 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3895 3896 return NO_ERROR; 3897} 3898 3899status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3900 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3901 3902 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3903 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3904 target); 3905 3906 if (!(target == TimedAudioTrack::LOCAL_TIME || 3907 target == TimedAudioTrack::COMMON_TIME)) { 3908 return BAD_VALUE; 3909 } 3910 3911 Mutex::Autolock lock(mMediaTimeTransformLock); 3912 mMediaTimeTransform = xform; 3913 mMediaTimeTransformTarget = target; 3914 mMediaTimeTransformValid = true; 3915 3916 return NO_ERROR; 3917} 3918 3919#define min(a, b) ((a) < (b) ? (a) : (b)) 3920 3921// implementation of getNextBuffer for tracks whose buffers have timestamps 3922status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3923 AudioBufferProvider::Buffer* buffer, int64_t pts) 3924{ 3925 if (pts == AudioBufferProvider::kInvalidPTS) { 3926 buffer->raw = 0; 3927 buffer->frameCount = 0; 3928 return INVALID_OPERATION; 3929 } 3930 3931 Mutex::Autolock _l(mTimedBufferQueueLock); 3932 3933 while (true) { 3934 3935 // if we have no timed buffers, then fail 3936 if (mTimedBufferQueue.isEmpty()) { 3937 buffer->raw = 0; 3938 buffer->frameCount = 0; 3939 return NOT_ENOUGH_DATA; 3940 } 3941 3942 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3943 3944 // calculate the PTS of the head of the timed buffer queue expressed in 3945 // local time 3946 int64_t headLocalPTS; 3947 { 3948 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3949 3950 assert(mMediaTimeTransformValid); 3951 3952 if (mMediaTimeTransform.a_to_b_denom == 0) { 3953 // the transform represents a pause, so yield silence 3954 timedYieldSilence(buffer->frameCount, buffer); 3955 return NO_ERROR; 3956 } 3957 3958 int64_t transformedPTS; 3959 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3960 &transformedPTS)) { 3961 // the transform failed. this shouldn't happen, but if it does 3962 // then just drop this buffer 3963 ALOGW("timedGetNextBuffer transform failed"); 3964 buffer->raw = 0; 3965 buffer->frameCount = 0; 3966 mTimedBufferQueue.removeAt(0); 3967 return NO_ERROR; 3968 } 3969 3970 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3971 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3972 &headLocalPTS)) { 3973 buffer->raw = 0; 3974 buffer->frameCount = 0; 3975 return INVALID_OPERATION; 3976 } 3977 } else { 3978 headLocalPTS = transformedPTS; 3979 } 3980 } 3981 3982 // adjust the head buffer's PTS to reflect the portion of the head buffer 3983 // that has already been consumed 3984 int64_t effectivePTS = headLocalPTS + 3985 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3986 3987 // Calculate the delta in samples between the head of the input buffer 3988 // queue and the start of the next output buffer that will be written. 3989 // If the transformation fails because of over or underflow, it means 3990 // that the sample's position in the output stream is so far out of 3991 // whack that it should just be dropped. 3992 int64_t sampleDelta; 3993 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3994 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3995 mTimedBufferQueue.removeAt(0); 3996 continue; 3997 } 3998 if (!mLocalTimeToSampleTransform.doForwardTransform( 3999 (effectivePTS - pts) << 32, &sampleDelta)) { 4000 ALOGV("*** too late during sample rate transform: dropped buffer"); 4001 mTimedBufferQueue.removeAt(0); 4002 continue; 4003 } 4004 4005 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4006 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4007 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4008 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4009 4010 // if the delta between the ideal placement for the next input sample and 4011 // the current output position is within this threshold, then we will 4012 // concatenate the next input samples to the previous output 4013 const int64_t kSampleContinuityThreshold = 4014 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4015 4016 // if this is the first buffer of audio that we're emitting from this track 4017 // then it should be almost exactly on time. 4018 const int64_t kSampleStartupThreshold = 1LL << 32; 4019 4020 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4021 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4022 // the next input is close enough to being on time, so concatenate it 4023 // with the last output 4024 timedYieldSamples(buffer); 4025 4026 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4027 return NO_ERROR; 4028 } else if (sampleDelta > 0) { 4029 // the gap between the current output position and the proper start of 4030 // the next input sample is too big, so fill it with silence 4031 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4032 4033 timedYieldSilence(framesUntilNextInput, buffer); 4034 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4035 return NO_ERROR; 4036 } else { 4037 // the next input sample is late 4038 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4039 size_t onTimeSamplePosition = 4040 head.position() + lateFrames * mCblk->frameSize; 4041 4042 if (onTimeSamplePosition > head.buffer()->size()) { 4043 // all the remaining samples in the head are too late, so 4044 // drop it and move on 4045 ALOGV("*** too late: dropped buffer"); 4046 mTimedBufferQueue.removeAt(0); 4047 continue; 4048 } else { 4049 // skip over the late samples 4050 head.setPosition(onTimeSamplePosition); 4051 4052 // yield the available samples 4053 timedYieldSamples(buffer); 4054 4055 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4056 return NO_ERROR; 4057 } 4058 } 4059 } 4060} 4061 4062// Yield samples from the timed buffer queue head up to the given output 4063// buffer's capacity. 4064// 4065// Caller must hold mTimedBufferQueueLock 4066void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4067 AudioBufferProvider::Buffer* buffer) { 4068 4069 const TimedBuffer& head = mTimedBufferQueue[0]; 4070 4071 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4072 head.position()); 4073 4074 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4075 mCblk->frameSize); 4076 size_t framesRequested = buffer->frameCount; 4077 buffer->frameCount = min(framesLeftInHead, framesRequested); 4078 4079 mTimedAudioOutputOnTime = true; 4080} 4081 4082// Yield samples of silence up to the given output buffer's capacity 4083// 4084// Caller must hold mTimedBufferQueueLock 4085void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4086 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4087 4088 // lazily allocate a buffer filled with silence 4089 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4090 delete [] mTimedSilenceBuffer; 4091 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4092 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4093 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4094 } 4095 4096 buffer->raw = mTimedSilenceBuffer; 4097 size_t framesRequested = buffer->frameCount; 4098 buffer->frameCount = min(numFrames, framesRequested); 4099 4100 mTimedAudioOutputOnTime = false; 4101} 4102 4103void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4104 AudioBufferProvider::Buffer* buffer) { 4105 4106 Mutex::Autolock _l(mTimedBufferQueueLock); 4107 4108 // If the buffer which was just released is part of the buffer at the head 4109 // of the queue, be sure to update the amt of the buffer which has been 4110 // consumed. If the buffer being returned is not part of the head of the 4111 // queue, its either because the buffer is part of the silence buffer, or 4112 // because the head of the timed queue was trimmed after the mixer called 4113 // getNextBuffer but before the mixer called releaseBuffer. 4114 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4115 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4116 4117 void* start = head.buffer()->pointer(); 4118 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4119 4120 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4121 head.setPosition(head.position() + 4122 (buffer->frameCount * mCblk->frameSize)); 4123 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4124 mTimedBufferQueue.removeAt(0); 4125 } 4126 } 4127 } 4128 4129 buffer->raw = 0; 4130 buffer->frameCount = 0; 4131} 4132 4133uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4134 Mutex::Autolock _l(mTimedBufferQueueLock); 4135 4136 uint32_t frames = 0; 4137 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4138 const TimedBuffer& tb = mTimedBufferQueue[i]; 4139 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4140 } 4141 4142 return frames; 4143} 4144 4145AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4146 : mPTS(0), mPosition(0) {} 4147 4148AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4149 const sp<IMemory>& buffer, int64_t pts) 4150 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4151 4152// ---------------------------------------------------------------------------- 4153 4154// RecordTrack constructor must be called with AudioFlinger::mLock held 4155AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4156 RecordThread *thread, 4157 const sp<Client>& client, 4158 uint32_t sampleRate, 4159 audio_format_t format, 4160 uint32_t channelMask, 4161 int frameCount, 4162 int sessionId) 4163 : TrackBase(thread, client, sampleRate, format, 4164 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4165 mOverflow(false) 4166{ 4167 if (mCblk != NULL) { 4168 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4169 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4170 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4171 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4172 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4173 } else { 4174 mCblk->frameSize = sizeof(int8_t); 4175 } 4176 } 4177} 4178 4179AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4180{ 4181 sp<ThreadBase> thread = mThread.promote(); 4182 if (thread != 0) { 4183 AudioSystem::releaseInput(thread->id()); 4184 } 4185} 4186 4187status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4188{ 4189 audio_track_cblk_t* cblk = this->cblk(); 4190 uint32_t framesAvail; 4191 uint32_t framesReq = buffer->frameCount; 4192 4193 // Check if last stepServer failed, try to step now 4194 if (mStepServerFailed) { 4195 if (!step()) goto getNextBuffer_exit; 4196 ALOGV("stepServer recovered"); 4197 mStepServerFailed = false; 4198 } 4199 4200 framesAvail = cblk->framesAvailable_l(); 4201 4202 if (CC_LIKELY(framesAvail)) { 4203 uint32_t s = cblk->server; 4204 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4205 4206 if (framesReq > framesAvail) { 4207 framesReq = framesAvail; 4208 } 4209 if (s + framesReq > bufferEnd) { 4210 framesReq = bufferEnd - s; 4211 } 4212 4213 buffer->raw = getBuffer(s, framesReq); 4214 if (buffer->raw == NULL) goto getNextBuffer_exit; 4215 4216 buffer->frameCount = framesReq; 4217 return NO_ERROR; 4218 } 4219 4220getNextBuffer_exit: 4221 buffer->raw = NULL; 4222 buffer->frameCount = 0; 4223 return NOT_ENOUGH_DATA; 4224} 4225 4226status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4227{ 4228 sp<ThreadBase> thread = mThread.promote(); 4229 if (thread != 0) { 4230 RecordThread *recordThread = (RecordThread *)thread.get(); 4231 return recordThread->start(this, tid); 4232 } else { 4233 return BAD_VALUE; 4234 } 4235} 4236 4237void AudioFlinger::RecordThread::RecordTrack::stop() 4238{ 4239 sp<ThreadBase> thread = mThread.promote(); 4240 if (thread != 0) { 4241 RecordThread *recordThread = (RecordThread *)thread.get(); 4242 recordThread->stop(this); 4243 TrackBase::reset(); 4244 // Force overerrun condition to avoid false overrun callback until first data is 4245 // read from buffer 4246 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4247 } 4248} 4249 4250void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4251{ 4252 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4253 (mClient == 0) ? getpid_cached : mClient->pid(), 4254 mFormat, 4255 mChannelMask, 4256 mSessionId, 4257 mFrameCount, 4258 mState, 4259 mCblk->sampleRate, 4260 mCblk->server, 4261 mCblk->user); 4262} 4263 4264 4265// ---------------------------------------------------------------------------- 4266 4267AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4268 PlaybackThread *playbackThread, 4269 DuplicatingThread *sourceThread, 4270 uint32_t sampleRate, 4271 audio_format_t format, 4272 uint32_t channelMask, 4273 int frameCount) 4274 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4275 mActive(false), mSourceThread(sourceThread) 4276{ 4277 4278 if (mCblk != NULL) { 4279 mCblk->flags |= CBLK_DIRECTION_OUT; 4280 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4281 mOutBuffer.frameCount = 0; 4282 playbackThread->mTracks.add(this); 4283 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4284 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4285 mCblk, mBuffer, mCblk->buffers, 4286 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4287 } else { 4288 ALOGW("Error creating output track on thread %p", playbackThread); 4289 } 4290} 4291 4292AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4293{ 4294 clearBufferQueue(); 4295} 4296 4297status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4298{ 4299 status_t status = Track::start(tid); 4300 if (status != NO_ERROR) { 4301 return status; 4302 } 4303 4304 mActive = true; 4305 mRetryCount = 127; 4306 return status; 4307} 4308 4309void AudioFlinger::PlaybackThread::OutputTrack::stop() 4310{ 4311 Track::stop(); 4312 clearBufferQueue(); 4313 mOutBuffer.frameCount = 0; 4314 mActive = false; 4315} 4316 4317bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4318{ 4319 Buffer *pInBuffer; 4320 Buffer inBuffer; 4321 uint32_t channelCount = mChannelCount; 4322 bool outputBufferFull = false; 4323 inBuffer.frameCount = frames; 4324 inBuffer.i16 = data; 4325 4326 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4327 4328 if (!mActive && frames != 0) { 4329 start(0); 4330 sp<ThreadBase> thread = mThread.promote(); 4331 if (thread != 0) { 4332 MixerThread *mixerThread = (MixerThread *)thread.get(); 4333 if (mCblk->frameCount > frames){ 4334 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4335 uint32_t startFrames = (mCblk->frameCount - frames); 4336 pInBuffer = new Buffer; 4337 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4338 pInBuffer->frameCount = startFrames; 4339 pInBuffer->i16 = pInBuffer->mBuffer; 4340 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4341 mBufferQueue.add(pInBuffer); 4342 } else { 4343 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4344 } 4345 } 4346 } 4347 } 4348 4349 while (waitTimeLeftMs) { 4350 // First write pending buffers, then new data 4351 if (mBufferQueue.size()) { 4352 pInBuffer = mBufferQueue.itemAt(0); 4353 } else { 4354 pInBuffer = &inBuffer; 4355 } 4356 4357 if (pInBuffer->frameCount == 0) { 4358 break; 4359 } 4360 4361 if (mOutBuffer.frameCount == 0) { 4362 mOutBuffer.frameCount = pInBuffer->frameCount; 4363 nsecs_t startTime = systemTime(); 4364 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4365 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4366 outputBufferFull = true; 4367 break; 4368 } 4369 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4370 if (waitTimeLeftMs >= waitTimeMs) { 4371 waitTimeLeftMs -= waitTimeMs; 4372 } else { 4373 waitTimeLeftMs = 0; 4374 } 4375 } 4376 4377 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4378 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4379 mCblk->stepUser(outFrames); 4380 pInBuffer->frameCount -= outFrames; 4381 pInBuffer->i16 += outFrames * channelCount; 4382 mOutBuffer.frameCount -= outFrames; 4383 mOutBuffer.i16 += outFrames * channelCount; 4384 4385 if (pInBuffer->frameCount == 0) { 4386 if (mBufferQueue.size()) { 4387 mBufferQueue.removeAt(0); 4388 delete [] pInBuffer->mBuffer; 4389 delete pInBuffer; 4390 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4391 } else { 4392 break; 4393 } 4394 } 4395 } 4396 4397 // If we could not write all frames, allocate a buffer and queue it for next time. 4398 if (inBuffer.frameCount) { 4399 sp<ThreadBase> thread = mThread.promote(); 4400 if (thread != 0 && !thread->standby()) { 4401 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4402 pInBuffer = new Buffer; 4403 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4404 pInBuffer->frameCount = inBuffer.frameCount; 4405 pInBuffer->i16 = pInBuffer->mBuffer; 4406 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4407 mBufferQueue.add(pInBuffer); 4408 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4409 } else { 4410 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4411 } 4412 } 4413 } 4414 4415 // Calling write() with a 0 length buffer, means that no more data will be written: 4416 // If no more buffers are pending, fill output track buffer to make sure it is started 4417 // by output mixer. 4418 if (frames == 0 && mBufferQueue.size() == 0) { 4419 if (mCblk->user < mCblk->frameCount) { 4420 frames = mCblk->frameCount - mCblk->user; 4421 pInBuffer = new Buffer; 4422 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4423 pInBuffer->frameCount = frames; 4424 pInBuffer->i16 = pInBuffer->mBuffer; 4425 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4426 mBufferQueue.add(pInBuffer); 4427 } else if (mActive) { 4428 stop(); 4429 } 4430 } 4431 4432 return outputBufferFull; 4433} 4434 4435status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4436{ 4437 int active; 4438 status_t result; 4439 audio_track_cblk_t* cblk = mCblk; 4440 uint32_t framesReq = buffer->frameCount; 4441 4442// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4443 buffer->frameCount = 0; 4444 4445 uint32_t framesAvail = cblk->framesAvailable(); 4446 4447 4448 if (framesAvail == 0) { 4449 Mutex::Autolock _l(cblk->lock); 4450 goto start_loop_here; 4451 while (framesAvail == 0) { 4452 active = mActive; 4453 if (CC_UNLIKELY(!active)) { 4454 ALOGV("Not active and NO_MORE_BUFFERS"); 4455 return NO_MORE_BUFFERS; 4456 } 4457 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4458 if (result != NO_ERROR) { 4459 return NO_MORE_BUFFERS; 4460 } 4461 // read the server count again 4462 start_loop_here: 4463 framesAvail = cblk->framesAvailable_l(); 4464 } 4465 } 4466 4467// if (framesAvail < framesReq) { 4468// return NO_MORE_BUFFERS; 4469// } 4470 4471 if (framesReq > framesAvail) { 4472 framesReq = framesAvail; 4473 } 4474 4475 uint32_t u = cblk->user; 4476 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4477 4478 if (u + framesReq > bufferEnd) { 4479 framesReq = bufferEnd - u; 4480 } 4481 4482 buffer->frameCount = framesReq; 4483 buffer->raw = (void *)cblk->buffer(u); 4484 return NO_ERROR; 4485} 4486 4487 4488void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4489{ 4490 size_t size = mBufferQueue.size(); 4491 4492 for (size_t i = 0; i < size; i++) { 4493 Buffer *pBuffer = mBufferQueue.itemAt(i); 4494 delete [] pBuffer->mBuffer; 4495 delete pBuffer; 4496 } 4497 mBufferQueue.clear(); 4498} 4499 4500// ---------------------------------------------------------------------------- 4501 4502AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4503 : RefBase(), 4504 mAudioFlinger(audioFlinger), 4505 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4506 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4507 mPid(pid), 4508 mTimedTrackCount(0) 4509{ 4510 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4511} 4512 4513// Client destructor must be called with AudioFlinger::mLock held 4514AudioFlinger::Client::~Client() 4515{ 4516 mAudioFlinger->removeClient_l(mPid); 4517} 4518 4519sp<MemoryDealer> AudioFlinger::Client::heap() const 4520{ 4521 return mMemoryDealer; 4522} 4523 4524// Reserve one of the limited slots for a timed audio track associated 4525// with this client 4526bool AudioFlinger::Client::reserveTimedTrack() 4527{ 4528 const int kMaxTimedTracksPerClient = 4; 4529 4530 Mutex::Autolock _l(mTimedTrackLock); 4531 4532 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4533 ALOGW("can not create timed track - pid %d has exceeded the limit", 4534 mPid); 4535 return false; 4536 } 4537 4538 mTimedTrackCount++; 4539 return true; 4540} 4541 4542// Release a slot for a timed audio track 4543void AudioFlinger::Client::releaseTimedTrack() 4544{ 4545 Mutex::Autolock _l(mTimedTrackLock); 4546 mTimedTrackCount--; 4547} 4548 4549// ---------------------------------------------------------------------------- 4550 4551AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4552 const sp<IAudioFlingerClient>& client, 4553 pid_t pid) 4554 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4555{ 4556} 4557 4558AudioFlinger::NotificationClient::~NotificationClient() 4559{ 4560} 4561 4562void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4563{ 4564 sp<NotificationClient> keep(this); 4565 mAudioFlinger->removeNotificationClient(mPid); 4566} 4567 4568// ---------------------------------------------------------------------------- 4569 4570AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4571 : BnAudioTrack(), 4572 mTrack(track) 4573{ 4574} 4575 4576AudioFlinger::TrackHandle::~TrackHandle() { 4577 // just stop the track on deletion, associated resources 4578 // will be freed from the main thread once all pending buffers have 4579 // been played. Unless it's not in the active track list, in which 4580 // case we free everything now... 4581 mTrack->destroy(); 4582} 4583 4584sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4585 return mTrack->getCblk(); 4586} 4587 4588status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4589 return mTrack->start(tid); 4590} 4591 4592void AudioFlinger::TrackHandle::stop() { 4593 mTrack->stop(); 4594} 4595 4596void AudioFlinger::TrackHandle::flush() { 4597 mTrack->flush(); 4598} 4599 4600void AudioFlinger::TrackHandle::mute(bool e) { 4601 mTrack->mute(e); 4602} 4603 4604void AudioFlinger::TrackHandle::pause() { 4605 mTrack->pause(); 4606} 4607 4608status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4609{ 4610 return mTrack->attachAuxEffect(EffectId); 4611} 4612 4613status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4614 sp<IMemory>* buffer) { 4615 if (!mTrack->isTimedTrack()) 4616 return INVALID_OPERATION; 4617 4618 PlaybackThread::TimedTrack* tt = 4619 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4620 return tt->allocateTimedBuffer(size, buffer); 4621} 4622 4623status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4624 int64_t pts) { 4625 if (!mTrack->isTimedTrack()) 4626 return INVALID_OPERATION; 4627 4628 PlaybackThread::TimedTrack* tt = 4629 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4630 return tt->queueTimedBuffer(buffer, pts); 4631} 4632 4633status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4634 const LinearTransform& xform, int target) { 4635 4636 if (!mTrack->isTimedTrack()) 4637 return INVALID_OPERATION; 4638 4639 PlaybackThread::TimedTrack* tt = 4640 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4641 return tt->setMediaTimeTransform( 4642 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4643} 4644 4645status_t AudioFlinger::TrackHandle::onTransact( 4646 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4647{ 4648 return BnAudioTrack::onTransact(code, data, reply, flags); 4649} 4650 4651// ---------------------------------------------------------------------------- 4652 4653sp<IAudioRecord> AudioFlinger::openRecord( 4654 pid_t pid, 4655 audio_io_handle_t input, 4656 uint32_t sampleRate, 4657 audio_format_t format, 4658 uint32_t channelMask, 4659 int frameCount, 4660 // FIXME dead, remove from IAudioFlinger 4661 uint32_t flags, 4662 int *sessionId, 4663 status_t *status) 4664{ 4665 sp<RecordThread::RecordTrack> recordTrack; 4666 sp<RecordHandle> recordHandle; 4667 sp<Client> client; 4668 status_t lStatus; 4669 RecordThread *thread; 4670 size_t inFrameCount; 4671 int lSessionId; 4672 4673 // check calling permissions 4674 if (!recordingAllowed()) { 4675 lStatus = PERMISSION_DENIED; 4676 goto Exit; 4677 } 4678 4679 // add client to list 4680 { // scope for mLock 4681 Mutex::Autolock _l(mLock); 4682 thread = checkRecordThread_l(input); 4683 if (thread == NULL) { 4684 lStatus = BAD_VALUE; 4685 goto Exit; 4686 } 4687 4688 client = registerPid_l(pid); 4689 4690 // If no audio session id is provided, create one here 4691 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4692 lSessionId = *sessionId; 4693 } else { 4694 lSessionId = nextUniqueId(); 4695 if (sessionId != NULL) { 4696 *sessionId = lSessionId; 4697 } 4698 } 4699 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4700 recordTrack = thread->createRecordTrack_l(client, 4701 sampleRate, 4702 format, 4703 channelMask, 4704 frameCount, 4705 lSessionId, 4706 &lStatus); 4707 } 4708 if (lStatus != NO_ERROR) { 4709 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4710 // destructor is called by the TrackBase destructor with mLock held 4711 client.clear(); 4712 recordTrack.clear(); 4713 goto Exit; 4714 } 4715 4716 // return to handle to client 4717 recordHandle = new RecordHandle(recordTrack); 4718 lStatus = NO_ERROR; 4719 4720Exit: 4721 if (status) { 4722 *status = lStatus; 4723 } 4724 return recordHandle; 4725} 4726 4727// ---------------------------------------------------------------------------- 4728 4729AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4730 : BnAudioRecord(), 4731 mRecordTrack(recordTrack) 4732{ 4733} 4734 4735AudioFlinger::RecordHandle::~RecordHandle() { 4736 stop(); 4737} 4738 4739sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4740 return mRecordTrack->getCblk(); 4741} 4742 4743status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4744 ALOGV("RecordHandle::start()"); 4745 return mRecordTrack->start(tid); 4746} 4747 4748void AudioFlinger::RecordHandle::stop() { 4749 ALOGV("RecordHandle::stop()"); 4750 mRecordTrack->stop(); 4751} 4752 4753status_t AudioFlinger::RecordHandle::onTransact( 4754 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4755{ 4756 return BnAudioRecord::onTransact(code, data, reply, flags); 4757} 4758 4759// ---------------------------------------------------------------------------- 4760 4761AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4762 AudioStreamIn *input, 4763 uint32_t sampleRate, 4764 uint32_t channels, 4765 audio_io_handle_t id, 4766 uint32_t device) : 4767 ThreadBase(audioFlinger, id, device, RECORD), 4768 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4769 // mRsmpInIndex and mInputBytes set by readInputParameters() 4770 mReqChannelCount(popcount(channels)), 4771 mReqSampleRate(sampleRate) 4772 // mBytesRead is only meaningful while active, and so is cleared in start() 4773 // (but might be better to also clear here for dump?) 4774{ 4775 snprintf(mName, kNameLength, "AudioIn_%d", id); 4776 4777 readInputParameters(); 4778} 4779 4780 4781AudioFlinger::RecordThread::~RecordThread() 4782{ 4783 delete[] mRsmpInBuffer; 4784 delete mResampler; 4785 delete[] mRsmpOutBuffer; 4786} 4787 4788void AudioFlinger::RecordThread::onFirstRef() 4789{ 4790 run(mName, PRIORITY_URGENT_AUDIO); 4791} 4792 4793status_t AudioFlinger::RecordThread::readyToRun() 4794{ 4795 status_t status = initCheck(); 4796 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4797 return status; 4798} 4799 4800bool AudioFlinger::RecordThread::threadLoop() 4801{ 4802 AudioBufferProvider::Buffer buffer; 4803 sp<RecordTrack> activeTrack; 4804 Vector< sp<EffectChain> > effectChains; 4805 4806 nsecs_t lastWarning = 0; 4807 4808 acquireWakeLock(); 4809 4810 // start recording 4811 while (!exitPending()) { 4812 4813 processConfigEvents(); 4814 4815 { // scope for mLock 4816 Mutex::Autolock _l(mLock); 4817 checkForNewParameters_l(); 4818 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4819 if (!mStandby) { 4820 mInput->stream->common.standby(&mInput->stream->common); 4821 mStandby = true; 4822 } 4823 4824 if (exitPending()) break; 4825 4826 releaseWakeLock_l(); 4827 ALOGV("RecordThread: loop stopping"); 4828 // go to sleep 4829 mWaitWorkCV.wait(mLock); 4830 ALOGV("RecordThread: loop starting"); 4831 acquireWakeLock_l(); 4832 continue; 4833 } 4834 if (mActiveTrack != 0) { 4835 if (mActiveTrack->mState == TrackBase::PAUSING) { 4836 if (!mStandby) { 4837 mInput->stream->common.standby(&mInput->stream->common); 4838 mStandby = true; 4839 } 4840 mActiveTrack.clear(); 4841 mStartStopCond.broadcast(); 4842 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4843 if (mReqChannelCount != mActiveTrack->channelCount()) { 4844 mActiveTrack.clear(); 4845 mStartStopCond.broadcast(); 4846 } else if (mBytesRead != 0) { 4847 // record start succeeds only if first read from audio input 4848 // succeeds 4849 if (mBytesRead > 0) { 4850 mActiveTrack->mState = TrackBase::ACTIVE; 4851 } else { 4852 mActiveTrack.clear(); 4853 } 4854 mStartStopCond.broadcast(); 4855 } 4856 mStandby = false; 4857 } 4858 } 4859 lockEffectChains_l(effectChains); 4860 } 4861 4862 if (mActiveTrack != 0) { 4863 if (mActiveTrack->mState != TrackBase::ACTIVE && 4864 mActiveTrack->mState != TrackBase::RESUMING) { 4865 unlockEffectChains(effectChains); 4866 usleep(kRecordThreadSleepUs); 4867 continue; 4868 } 4869 for (size_t i = 0; i < effectChains.size(); i ++) { 4870 effectChains[i]->process_l(); 4871 } 4872 4873 buffer.frameCount = mFrameCount; 4874 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4875 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4876 size_t framesOut = buffer.frameCount; 4877 if (mResampler == NULL) { 4878 // no resampling 4879 while (framesOut) { 4880 size_t framesIn = mFrameCount - mRsmpInIndex; 4881 if (framesIn) { 4882 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4883 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4884 if (framesIn > framesOut) 4885 framesIn = framesOut; 4886 mRsmpInIndex += framesIn; 4887 framesOut -= framesIn; 4888 if ((int)mChannelCount == mReqChannelCount || 4889 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4890 memcpy(dst, src, framesIn * mFrameSize); 4891 } else { 4892 int16_t *src16 = (int16_t *)src; 4893 int16_t *dst16 = (int16_t *)dst; 4894 if (mChannelCount == 1) { 4895 while (framesIn--) { 4896 *dst16++ = *src16; 4897 *dst16++ = *src16++; 4898 } 4899 } else { 4900 while (framesIn--) { 4901 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4902 src16 += 2; 4903 } 4904 } 4905 } 4906 } 4907 if (framesOut && mFrameCount == mRsmpInIndex) { 4908 if (framesOut == mFrameCount && 4909 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4910 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4911 framesOut = 0; 4912 } else { 4913 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4914 mRsmpInIndex = 0; 4915 } 4916 if (mBytesRead < 0) { 4917 ALOGE("Error reading audio input"); 4918 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4919 // Force input into standby so that it tries to 4920 // recover at next read attempt 4921 mInput->stream->common.standby(&mInput->stream->common); 4922 usleep(kRecordThreadSleepUs); 4923 } 4924 mRsmpInIndex = mFrameCount; 4925 framesOut = 0; 4926 buffer.frameCount = 0; 4927 } 4928 } 4929 } 4930 } else { 4931 // resampling 4932 4933 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4934 // alter output frame count as if we were expecting stereo samples 4935 if (mChannelCount == 1 && mReqChannelCount == 1) { 4936 framesOut >>= 1; 4937 } 4938 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4939 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4940 // are 32 bit aligned which should be always true. 4941 if (mChannelCount == 2 && mReqChannelCount == 1) { 4942 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4943 // the resampler always outputs stereo samples: do post stereo to mono conversion 4944 int16_t *src = (int16_t *)mRsmpOutBuffer; 4945 int16_t *dst = buffer.i16; 4946 while (framesOut--) { 4947 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4948 src += 2; 4949 } 4950 } else { 4951 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4952 } 4953 4954 } 4955 mActiveTrack->releaseBuffer(&buffer); 4956 mActiveTrack->overflow(); 4957 } 4958 // client isn't retrieving buffers fast enough 4959 else { 4960 if (!mActiveTrack->setOverflow()) { 4961 nsecs_t now = systemTime(); 4962 if ((now - lastWarning) > kWarningThrottleNs) { 4963 ALOGW("RecordThread: buffer overflow"); 4964 lastWarning = now; 4965 } 4966 } 4967 // Release the processor for a while before asking for a new buffer. 4968 // This will give the application more chance to read from the buffer and 4969 // clear the overflow. 4970 usleep(kRecordThreadSleepUs); 4971 } 4972 } 4973 // enable changes in effect chain 4974 unlockEffectChains(effectChains); 4975 effectChains.clear(); 4976 } 4977 4978 if (!mStandby) { 4979 mInput->stream->common.standby(&mInput->stream->common); 4980 } 4981 mActiveTrack.clear(); 4982 4983 mStartStopCond.broadcast(); 4984 4985 releaseWakeLock(); 4986 4987 ALOGV("RecordThread %p exiting", this); 4988 return false; 4989} 4990 4991 4992sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4993 const sp<AudioFlinger::Client>& client, 4994 uint32_t sampleRate, 4995 audio_format_t format, 4996 int channelMask, 4997 int frameCount, 4998 int sessionId, 4999 status_t *status) 5000{ 5001 sp<RecordTrack> track; 5002 status_t lStatus; 5003 5004 lStatus = initCheck(); 5005 if (lStatus != NO_ERROR) { 5006 ALOGE("Audio driver not initialized."); 5007 goto Exit; 5008 } 5009 5010 { // scope for mLock 5011 Mutex::Autolock _l(mLock); 5012 5013 track = new RecordTrack(this, client, sampleRate, 5014 format, channelMask, frameCount, sessionId); 5015 5016 if (track->getCblk() == 0) { 5017 lStatus = NO_MEMORY; 5018 goto Exit; 5019 } 5020 5021 mTrack = track.get(); 5022 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5023 bool suspend = audio_is_bluetooth_sco_device( 5024 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5025 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5026 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5027 } 5028 lStatus = NO_ERROR; 5029 5030Exit: 5031 if (status) { 5032 *status = lStatus; 5033 } 5034 return track; 5035} 5036 5037status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5038{ 5039 ALOGV("RecordThread::start tid=%d", tid); 5040 sp <ThreadBase> strongMe = this; 5041 status_t status = NO_ERROR; 5042 { 5043 AutoMutex lock(mLock); 5044 if (mActiveTrack != 0) { 5045 if (recordTrack != mActiveTrack.get()) { 5046 status = -EBUSY; 5047 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5048 mActiveTrack->mState = TrackBase::ACTIVE; 5049 } 5050 return status; 5051 } 5052 5053 recordTrack->mState = TrackBase::IDLE; 5054 mActiveTrack = recordTrack; 5055 mLock.unlock(); 5056 status_t status = AudioSystem::startInput(mId); 5057 mLock.lock(); 5058 if (status != NO_ERROR) { 5059 mActiveTrack.clear(); 5060 return status; 5061 } 5062 mRsmpInIndex = mFrameCount; 5063 mBytesRead = 0; 5064 if (mResampler != NULL) { 5065 mResampler->reset(); 5066 } 5067 mActiveTrack->mState = TrackBase::RESUMING; 5068 // signal thread to start 5069 ALOGV("Signal record thread"); 5070 mWaitWorkCV.signal(); 5071 // do not wait for mStartStopCond if exiting 5072 if (exitPending()) { 5073 mActiveTrack.clear(); 5074 status = INVALID_OPERATION; 5075 goto startError; 5076 } 5077 mStartStopCond.wait(mLock); 5078 if (mActiveTrack == 0) { 5079 ALOGV("Record failed to start"); 5080 status = BAD_VALUE; 5081 goto startError; 5082 } 5083 ALOGV("Record started OK"); 5084 return status; 5085 } 5086startError: 5087 AudioSystem::stopInput(mId); 5088 return status; 5089} 5090 5091void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5092 ALOGV("RecordThread::stop"); 5093 sp <ThreadBase> strongMe = this; 5094 { 5095 AutoMutex lock(mLock); 5096 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5097 mActiveTrack->mState = TrackBase::PAUSING; 5098 // do not wait for mStartStopCond if exiting 5099 if (exitPending()) { 5100 return; 5101 } 5102 mStartStopCond.wait(mLock); 5103 // if we have been restarted, recordTrack == mActiveTrack.get() here 5104 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5105 mLock.unlock(); 5106 AudioSystem::stopInput(mId); 5107 mLock.lock(); 5108 ALOGV("Record stopped OK"); 5109 } 5110 } 5111 } 5112} 5113 5114status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5115{ 5116 const size_t SIZE = 256; 5117 char buffer[SIZE]; 5118 String8 result; 5119 5120 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5121 result.append(buffer); 5122 5123 if (mActiveTrack != 0) { 5124 result.append("Active Track:\n"); 5125 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5126 mActiveTrack->dump(buffer, SIZE); 5127 result.append(buffer); 5128 5129 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5130 result.append(buffer); 5131 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5132 result.append(buffer); 5133 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5134 result.append(buffer); 5135 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5136 result.append(buffer); 5137 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5138 result.append(buffer); 5139 5140 5141 } else { 5142 result.append("No record client\n"); 5143 } 5144 write(fd, result.string(), result.size()); 5145 5146 dumpBase(fd, args); 5147 dumpEffectChains(fd, args); 5148 5149 return NO_ERROR; 5150} 5151 5152status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5153{ 5154 size_t framesReq = buffer->frameCount; 5155 size_t framesReady = mFrameCount - mRsmpInIndex; 5156 int channelCount; 5157 5158 if (framesReady == 0) { 5159 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5160 if (mBytesRead < 0) { 5161 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5162 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5163 // Force input into standby so that it tries to 5164 // recover at next read attempt 5165 mInput->stream->common.standby(&mInput->stream->common); 5166 usleep(kRecordThreadSleepUs); 5167 } 5168 buffer->raw = NULL; 5169 buffer->frameCount = 0; 5170 return NOT_ENOUGH_DATA; 5171 } 5172 mRsmpInIndex = 0; 5173 framesReady = mFrameCount; 5174 } 5175 5176 if (framesReq > framesReady) { 5177 framesReq = framesReady; 5178 } 5179 5180 if (mChannelCount == 1 && mReqChannelCount == 2) { 5181 channelCount = 1; 5182 } else { 5183 channelCount = 2; 5184 } 5185 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5186 buffer->frameCount = framesReq; 5187 return NO_ERROR; 5188} 5189 5190void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5191{ 5192 mRsmpInIndex += buffer->frameCount; 5193 buffer->frameCount = 0; 5194} 5195 5196bool AudioFlinger::RecordThread::checkForNewParameters_l() 5197{ 5198 bool reconfig = false; 5199 5200 while (!mNewParameters.isEmpty()) { 5201 status_t status = NO_ERROR; 5202 String8 keyValuePair = mNewParameters[0]; 5203 AudioParameter param = AudioParameter(keyValuePair); 5204 int value; 5205 audio_format_t reqFormat = mFormat; 5206 int reqSamplingRate = mReqSampleRate; 5207 int reqChannelCount = mReqChannelCount; 5208 5209 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5210 reqSamplingRate = value; 5211 reconfig = true; 5212 } 5213 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5214 reqFormat = (audio_format_t) value; 5215 reconfig = true; 5216 } 5217 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5218 reqChannelCount = popcount(value); 5219 reconfig = true; 5220 } 5221 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5222 // do not accept frame count changes if tracks are open as the track buffer 5223 // size depends on frame count and correct behavior would not be guaranteed 5224 // if frame count is changed after track creation 5225 if (mActiveTrack != 0) { 5226 status = INVALID_OPERATION; 5227 } else { 5228 reconfig = true; 5229 } 5230 } 5231 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5232 // forward device change to effects that have requested to be 5233 // aware of attached audio device. 5234 for (size_t i = 0; i < mEffectChains.size(); i++) { 5235 mEffectChains[i]->setDevice_l(value); 5236 } 5237 // store input device and output device but do not forward output device to audio HAL. 5238 // Note that status is ignored by the caller for output device 5239 // (see AudioFlinger::setParameters() 5240 if (value & AUDIO_DEVICE_OUT_ALL) { 5241 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5242 status = BAD_VALUE; 5243 } else { 5244 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5245 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5246 if (mTrack != NULL) { 5247 bool suspend = audio_is_bluetooth_sco_device( 5248 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5249 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5250 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5251 } 5252 } 5253 mDevice |= (uint32_t)value; 5254 } 5255 if (status == NO_ERROR) { 5256 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5257 if (status == INVALID_OPERATION) { 5258 mInput->stream->common.standby(&mInput->stream->common); 5259 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5260 } 5261 if (reconfig) { 5262 if (status == BAD_VALUE && 5263 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5264 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5265 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5266 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5267 (reqChannelCount < 3)) { 5268 status = NO_ERROR; 5269 } 5270 if (status == NO_ERROR) { 5271 readInputParameters(); 5272 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5273 } 5274 } 5275 } 5276 5277 mNewParameters.removeAt(0); 5278 5279 mParamStatus = status; 5280 mParamCond.signal(); 5281 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5282 // already timed out waiting for the status and will never signal the condition. 5283 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5284 } 5285 return reconfig; 5286} 5287 5288String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5289{ 5290 char *s; 5291 String8 out_s8 = String8(); 5292 5293 Mutex::Autolock _l(mLock); 5294 if (initCheck() != NO_ERROR) { 5295 return out_s8; 5296 } 5297 5298 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5299 out_s8 = String8(s); 5300 free(s); 5301 return out_s8; 5302} 5303 5304void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5305 AudioSystem::OutputDescriptor desc; 5306 void *param2 = NULL; 5307 5308 switch (event) { 5309 case AudioSystem::INPUT_OPENED: 5310 case AudioSystem::INPUT_CONFIG_CHANGED: 5311 desc.channels = mChannelMask; 5312 desc.samplingRate = mSampleRate; 5313 desc.format = mFormat; 5314 desc.frameCount = mFrameCount; 5315 desc.latency = 0; 5316 param2 = &desc; 5317 break; 5318 5319 case AudioSystem::INPUT_CLOSED: 5320 default: 5321 break; 5322 } 5323 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5324} 5325 5326void AudioFlinger::RecordThread::readInputParameters() 5327{ 5328 delete mRsmpInBuffer; 5329 // mRsmpInBuffer is always assigned a new[] below 5330 delete mRsmpOutBuffer; 5331 mRsmpOutBuffer = NULL; 5332 delete mResampler; 5333 mResampler = NULL; 5334 5335 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5336 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5337 mChannelCount = (uint16_t)popcount(mChannelMask); 5338 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5339 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5340 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5341 mFrameCount = mInputBytes / mFrameSize; 5342 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5343 5344 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5345 { 5346 int channelCount; 5347 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5348 // stereo to mono post process as the resampler always outputs stereo. 5349 if (mChannelCount == 1 && mReqChannelCount == 2) { 5350 channelCount = 1; 5351 } else { 5352 channelCount = 2; 5353 } 5354 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5355 mResampler->setSampleRate(mSampleRate); 5356 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5357 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5358 5359 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5360 if (mChannelCount == 1 && mReqChannelCount == 1) { 5361 mFrameCount >>= 1; 5362 } 5363 5364 } 5365 mRsmpInIndex = mFrameCount; 5366} 5367 5368unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5369{ 5370 Mutex::Autolock _l(mLock); 5371 if (initCheck() != NO_ERROR) { 5372 return 0; 5373 } 5374 5375 return mInput->stream->get_input_frames_lost(mInput->stream); 5376} 5377 5378uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5379{ 5380 Mutex::Autolock _l(mLock); 5381 uint32_t result = 0; 5382 if (getEffectChain_l(sessionId) != 0) { 5383 result = EFFECT_SESSION; 5384 } 5385 5386 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5387 result |= TRACK_SESSION; 5388 } 5389 5390 return result; 5391} 5392 5393AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5394{ 5395 Mutex::Autolock _l(mLock); 5396 return mTrack; 5397} 5398 5399AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5400{ 5401 Mutex::Autolock _l(mLock); 5402 return mInput; 5403} 5404 5405AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5406{ 5407 Mutex::Autolock _l(mLock); 5408 AudioStreamIn *input = mInput; 5409 mInput = NULL; 5410 return input; 5411} 5412 5413// this method must always be called either with ThreadBase mLock held or inside the thread loop 5414audio_stream_t* AudioFlinger::RecordThread::stream() 5415{ 5416 if (mInput == NULL) { 5417 return NULL; 5418 } 5419 return &mInput->stream->common; 5420} 5421 5422 5423// ---------------------------------------------------------------------------- 5424 5425audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5426 uint32_t *pSamplingRate, 5427 audio_format_t *pFormat, 5428 uint32_t *pChannels, 5429 uint32_t *pLatencyMs, 5430 uint32_t flags) 5431{ 5432 status_t status; 5433 PlaybackThread *thread = NULL; 5434 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5435 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5436 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5437 uint32_t channels = pChannels ? *pChannels : 0; 5438 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5439 audio_stream_out_t *outStream; 5440 audio_hw_device_t *outHwDev; 5441 5442 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5443 pDevices ? *pDevices : 0, 5444 samplingRate, 5445 format, 5446 channels, 5447 flags); 5448 5449 if (pDevices == NULL || *pDevices == 0) { 5450 return 0; 5451 } 5452 5453 Mutex::Autolock _l(mLock); 5454 5455 outHwDev = findSuitableHwDev_l(*pDevices); 5456 if (outHwDev == NULL) 5457 return 0; 5458 5459 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5460 &channels, &samplingRate, &outStream); 5461 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5462 outStream, 5463 samplingRate, 5464 format, 5465 channels, 5466 status); 5467 5468 mHardwareStatus = AUDIO_HW_IDLE; 5469 if (outStream != NULL) { 5470 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5471 audio_io_handle_t id = nextUniqueId(); 5472 5473 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5474 (format != AUDIO_FORMAT_PCM_16_BIT) || 5475 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5476 thread = new DirectOutputThread(this, output, id, *pDevices); 5477 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5478 } else { 5479 thread = new MixerThread(this, output, id, *pDevices); 5480 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5481 } 5482 mPlaybackThreads.add(id, thread); 5483 5484 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5485 if (pFormat != NULL) *pFormat = format; 5486 if (pChannels != NULL) *pChannels = channels; 5487 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5488 5489 // notify client processes of the new output creation 5490 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5491 return id; 5492 } 5493 5494 return 0; 5495} 5496 5497audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5498 audio_io_handle_t output2) 5499{ 5500 Mutex::Autolock _l(mLock); 5501 MixerThread *thread1 = checkMixerThread_l(output1); 5502 MixerThread *thread2 = checkMixerThread_l(output2); 5503 5504 if (thread1 == NULL || thread2 == NULL) { 5505 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5506 return 0; 5507 } 5508 5509 audio_io_handle_t id = nextUniqueId(); 5510 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5511 thread->addOutputTrack(thread2); 5512 mPlaybackThreads.add(id, thread); 5513 // notify client processes of the new output creation 5514 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5515 return id; 5516} 5517 5518status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5519{ 5520 // keep strong reference on the playback thread so that 5521 // it is not destroyed while exit() is executed 5522 sp <PlaybackThread> thread; 5523 { 5524 Mutex::Autolock _l(mLock); 5525 thread = checkPlaybackThread_l(output); 5526 if (thread == NULL) { 5527 return BAD_VALUE; 5528 } 5529 5530 ALOGV("closeOutput() %d", output); 5531 5532 if (thread->type() == ThreadBase::MIXER) { 5533 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5534 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5535 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5536 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5537 } 5538 } 5539 } 5540 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5541 mPlaybackThreads.removeItem(output); 5542 } 5543 thread->exit(); 5544 // The thread entity (active unit of execution) is no longer running here, 5545 // but the ThreadBase container still exists. 5546 5547 if (thread->type() != ThreadBase::DUPLICATING) { 5548 AudioStreamOut *out = thread->clearOutput(); 5549 assert(out != NULL); 5550 // from now on thread->mOutput is NULL 5551 out->hwDev->close_output_stream(out->hwDev, out->stream); 5552 delete out; 5553 } 5554 return NO_ERROR; 5555} 5556 5557status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5558{ 5559 Mutex::Autolock _l(mLock); 5560 PlaybackThread *thread = checkPlaybackThread_l(output); 5561 5562 if (thread == NULL) { 5563 return BAD_VALUE; 5564 } 5565 5566 ALOGV("suspendOutput() %d", output); 5567 thread->suspend(); 5568 5569 return NO_ERROR; 5570} 5571 5572status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5573{ 5574 Mutex::Autolock _l(mLock); 5575 PlaybackThread *thread = checkPlaybackThread_l(output); 5576 5577 if (thread == NULL) { 5578 return BAD_VALUE; 5579 } 5580 5581 ALOGV("restoreOutput() %d", output); 5582 5583 thread->restore(); 5584 5585 return NO_ERROR; 5586} 5587 5588audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5589 uint32_t *pSamplingRate, 5590 audio_format_t *pFormat, 5591 uint32_t *pChannels, 5592 audio_in_acoustics_t acoustics) 5593{ 5594 status_t status; 5595 RecordThread *thread = NULL; 5596 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5597 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5598 uint32_t channels = pChannels ? *pChannels : 0; 5599 uint32_t reqSamplingRate = samplingRate; 5600 audio_format_t reqFormat = format; 5601 uint32_t reqChannels = channels; 5602 audio_stream_in_t *inStream; 5603 audio_hw_device_t *inHwDev; 5604 5605 if (pDevices == NULL || *pDevices == 0) { 5606 return 0; 5607 } 5608 5609 Mutex::Autolock _l(mLock); 5610 5611 inHwDev = findSuitableHwDev_l(*pDevices); 5612 if (inHwDev == NULL) 5613 return 0; 5614 5615 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5616 &channels, &samplingRate, 5617 acoustics, 5618 &inStream); 5619 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5620 inStream, 5621 samplingRate, 5622 format, 5623 channels, 5624 acoustics, 5625 status); 5626 5627 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5628 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5629 // or stereo to mono conversions on 16 bit PCM inputs. 5630 if (inStream == NULL && status == BAD_VALUE && 5631 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5632 (samplingRate <= 2 * reqSamplingRate) && 5633 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5634 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5635 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5636 &channels, &samplingRate, 5637 acoustics, 5638 &inStream); 5639 } 5640 5641 if (inStream != NULL) { 5642 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5643 5644 audio_io_handle_t id = nextUniqueId(); 5645 // Start record thread 5646 // RecorThread require both input and output device indication to forward to audio 5647 // pre processing modules 5648 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5649 thread = new RecordThread(this, 5650 input, 5651 reqSamplingRate, 5652 reqChannels, 5653 id, 5654 device); 5655 mRecordThreads.add(id, thread); 5656 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5657 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5658 if (pFormat != NULL) *pFormat = format; 5659 if (pChannels != NULL) *pChannels = reqChannels; 5660 5661 input->stream->common.standby(&input->stream->common); 5662 5663 // notify client processes of the new input creation 5664 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5665 return id; 5666 } 5667 5668 return 0; 5669} 5670 5671status_t AudioFlinger::closeInput(audio_io_handle_t input) 5672{ 5673 // keep strong reference on the record thread so that 5674 // it is not destroyed while exit() is executed 5675 sp <RecordThread> thread; 5676 { 5677 Mutex::Autolock _l(mLock); 5678 thread = checkRecordThread_l(input); 5679 if (thread == NULL) { 5680 return BAD_VALUE; 5681 } 5682 5683 ALOGV("closeInput() %d", input); 5684 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5685 mRecordThreads.removeItem(input); 5686 } 5687 thread->exit(); 5688 // The thread entity (active unit of execution) is no longer running here, 5689 // but the ThreadBase container still exists. 5690 5691 AudioStreamIn *in = thread->clearInput(); 5692 assert(in != NULL); 5693 // from now on thread->mInput is NULL 5694 in->hwDev->close_input_stream(in->hwDev, in->stream); 5695 delete in; 5696 5697 return NO_ERROR; 5698} 5699 5700status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5701{ 5702 Mutex::Autolock _l(mLock); 5703 MixerThread *dstThread = checkMixerThread_l(output); 5704 if (dstThread == NULL) { 5705 ALOGW("setStreamOutput() bad output id %d", output); 5706 return BAD_VALUE; 5707 } 5708 5709 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5710 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5711 5712 dstThread->setStreamValid(stream, true); 5713 5714 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5715 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5716 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5717 MixerThread *srcThread = (MixerThread *)thread; 5718 srcThread->setStreamValid(stream, false); 5719 srcThread->invalidateTracks(stream); 5720 } 5721 } 5722 5723 return NO_ERROR; 5724} 5725 5726 5727int AudioFlinger::newAudioSessionId() 5728{ 5729 return nextUniqueId(); 5730} 5731 5732void AudioFlinger::acquireAudioSessionId(int audioSession) 5733{ 5734 Mutex::Autolock _l(mLock); 5735 pid_t caller = IPCThreadState::self()->getCallingPid(); 5736 ALOGV("acquiring %d from %d", audioSession, caller); 5737 size_t num = mAudioSessionRefs.size(); 5738 for (size_t i = 0; i< num; i++) { 5739 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5740 if (ref->sessionid == audioSession && ref->pid == caller) { 5741 ref->cnt++; 5742 ALOGV(" incremented refcount to %d", ref->cnt); 5743 return; 5744 } 5745 } 5746 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5747 ALOGV(" added new entry for %d", audioSession); 5748} 5749 5750void AudioFlinger::releaseAudioSessionId(int audioSession) 5751{ 5752 Mutex::Autolock _l(mLock); 5753 pid_t caller = IPCThreadState::self()->getCallingPid(); 5754 ALOGV("releasing %d from %d", audioSession, caller); 5755 size_t num = mAudioSessionRefs.size(); 5756 for (size_t i = 0; i< num; i++) { 5757 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5758 if (ref->sessionid == audioSession && ref->pid == caller) { 5759 ref->cnt--; 5760 ALOGV(" decremented refcount to %d", ref->cnt); 5761 if (ref->cnt == 0) { 5762 mAudioSessionRefs.removeAt(i); 5763 delete ref; 5764 purgeStaleEffects_l(); 5765 } 5766 return; 5767 } 5768 } 5769 ALOGW("session id %d not found for pid %d", audioSession, caller); 5770} 5771 5772void AudioFlinger::purgeStaleEffects_l() { 5773 5774 ALOGV("purging stale effects"); 5775 5776 Vector< sp<EffectChain> > chains; 5777 5778 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5779 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5780 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5781 sp<EffectChain> ec = t->mEffectChains[j]; 5782 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5783 chains.push(ec); 5784 } 5785 } 5786 } 5787 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5788 sp<RecordThread> t = mRecordThreads.valueAt(i); 5789 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5790 sp<EffectChain> ec = t->mEffectChains[j]; 5791 chains.push(ec); 5792 } 5793 } 5794 5795 for (size_t i = 0; i < chains.size(); i++) { 5796 sp<EffectChain> ec = chains[i]; 5797 int sessionid = ec->sessionId(); 5798 sp<ThreadBase> t = ec->mThread.promote(); 5799 if (t == 0) { 5800 continue; 5801 } 5802 size_t numsessionrefs = mAudioSessionRefs.size(); 5803 bool found = false; 5804 for (size_t k = 0; k < numsessionrefs; k++) { 5805 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5806 if (ref->sessionid == sessionid) { 5807 ALOGV(" session %d still exists for %d with %d refs", 5808 sessionid, ref->pid, ref->cnt); 5809 found = true; 5810 break; 5811 } 5812 } 5813 if (!found) { 5814 // remove all effects from the chain 5815 while (ec->mEffects.size()) { 5816 sp<EffectModule> effect = ec->mEffects[0]; 5817 effect->unPin(); 5818 Mutex::Autolock _l (t->mLock); 5819 t->removeEffect_l(effect); 5820 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5821 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5822 if (handle != 0) { 5823 handle->mEffect.clear(); 5824 if (handle->mHasControl && handle->mEnabled) { 5825 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5826 } 5827 } 5828 } 5829 AudioSystem::unregisterEffect(effect->id()); 5830 } 5831 } 5832 } 5833 return; 5834} 5835 5836// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5837AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5838{ 5839 return mPlaybackThreads.valueFor(output).get(); 5840} 5841 5842// checkMixerThread_l() must be called with AudioFlinger::mLock held 5843AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5844{ 5845 PlaybackThread *thread = checkPlaybackThread_l(output); 5846 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5847} 5848 5849// checkRecordThread_l() must be called with AudioFlinger::mLock held 5850AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5851{ 5852 return mRecordThreads.valueFor(input).get(); 5853} 5854 5855uint32_t AudioFlinger::nextUniqueId() 5856{ 5857 return android_atomic_inc(&mNextUniqueId); 5858} 5859 5860AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5861{ 5862 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5863 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5864 AudioStreamOut *output = thread->getOutput(); 5865 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5866 return thread; 5867 } 5868 } 5869 return NULL; 5870} 5871 5872uint32_t AudioFlinger::primaryOutputDevice_l() 5873{ 5874 PlaybackThread *thread = primaryPlaybackThread_l(); 5875 5876 if (thread == NULL) { 5877 return 0; 5878 } 5879 5880 return thread->device(); 5881} 5882 5883 5884// ---------------------------------------------------------------------------- 5885// Effect management 5886// ---------------------------------------------------------------------------- 5887 5888 5889status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5890{ 5891 Mutex::Autolock _l(mLock); 5892 return EffectQueryNumberEffects(numEffects); 5893} 5894 5895status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5896{ 5897 Mutex::Autolock _l(mLock); 5898 return EffectQueryEffect(index, descriptor); 5899} 5900 5901status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5902 effect_descriptor_t *descriptor) const 5903{ 5904 Mutex::Autolock _l(mLock); 5905 return EffectGetDescriptor(pUuid, descriptor); 5906} 5907 5908 5909sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5910 effect_descriptor_t *pDesc, 5911 const sp<IEffectClient>& effectClient, 5912 int32_t priority, 5913 audio_io_handle_t io, 5914 int sessionId, 5915 status_t *status, 5916 int *id, 5917 int *enabled) 5918{ 5919 status_t lStatus = NO_ERROR; 5920 sp<EffectHandle> handle; 5921 effect_descriptor_t desc; 5922 5923 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5924 pid, effectClient.get(), priority, sessionId, io); 5925 5926 if (pDesc == NULL) { 5927 lStatus = BAD_VALUE; 5928 goto Exit; 5929 } 5930 5931 // check audio settings permission for global effects 5932 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5933 lStatus = PERMISSION_DENIED; 5934 goto Exit; 5935 } 5936 5937 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5938 // that can only be created by audio policy manager (running in same process) 5939 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5940 lStatus = PERMISSION_DENIED; 5941 goto Exit; 5942 } 5943 5944 if (io == 0) { 5945 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5946 // output must be specified by AudioPolicyManager when using session 5947 // AUDIO_SESSION_OUTPUT_STAGE 5948 lStatus = BAD_VALUE; 5949 goto Exit; 5950 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5951 // if the output returned by getOutputForEffect() is removed before we lock the 5952 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5953 // and we will exit safely 5954 io = AudioSystem::getOutputForEffect(&desc); 5955 } 5956 } 5957 5958 { 5959 Mutex::Autolock _l(mLock); 5960 5961 5962 if (!EffectIsNullUuid(&pDesc->uuid)) { 5963 // if uuid is specified, request effect descriptor 5964 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5965 if (lStatus < 0) { 5966 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5967 goto Exit; 5968 } 5969 } else { 5970 // if uuid is not specified, look for an available implementation 5971 // of the required type in effect factory 5972 if (EffectIsNullUuid(&pDesc->type)) { 5973 ALOGW("createEffect() no effect type"); 5974 lStatus = BAD_VALUE; 5975 goto Exit; 5976 } 5977 uint32_t numEffects = 0; 5978 effect_descriptor_t d; 5979 d.flags = 0; // prevent compiler warning 5980 bool found = false; 5981 5982 lStatus = EffectQueryNumberEffects(&numEffects); 5983 if (lStatus < 0) { 5984 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5985 goto Exit; 5986 } 5987 for (uint32_t i = 0; i < numEffects; i++) { 5988 lStatus = EffectQueryEffect(i, &desc); 5989 if (lStatus < 0) { 5990 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5991 continue; 5992 } 5993 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5994 // If matching type found save effect descriptor. If the session is 5995 // 0 and the effect is not auxiliary, continue enumeration in case 5996 // an auxiliary version of this effect type is available 5997 found = true; 5998 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5999 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6000 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6001 break; 6002 } 6003 } 6004 } 6005 if (!found) { 6006 lStatus = BAD_VALUE; 6007 ALOGW("createEffect() effect not found"); 6008 goto Exit; 6009 } 6010 // For same effect type, chose auxiliary version over insert version if 6011 // connect to output mix (Compliance to OpenSL ES) 6012 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6013 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6014 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6015 } 6016 } 6017 6018 // Do not allow auxiliary effects on a session different from 0 (output mix) 6019 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6020 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6021 lStatus = INVALID_OPERATION; 6022 goto Exit; 6023 } 6024 6025 // check recording permission for visualizer 6026 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6027 !recordingAllowed()) { 6028 lStatus = PERMISSION_DENIED; 6029 goto Exit; 6030 } 6031 6032 // return effect descriptor 6033 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6034 6035 // If output is not specified try to find a matching audio session ID in one of the 6036 // output threads. 6037 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6038 // because of code checking output when entering the function. 6039 // Note: io is never 0 when creating an effect on an input 6040 if (io == 0) { 6041 // look for the thread where the specified audio session is present 6042 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6043 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6044 io = mPlaybackThreads.keyAt(i); 6045 break; 6046 } 6047 } 6048 if (io == 0) { 6049 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6050 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6051 io = mRecordThreads.keyAt(i); 6052 break; 6053 } 6054 } 6055 } 6056 // If no output thread contains the requested session ID, default to 6057 // first output. The effect chain will be moved to the correct output 6058 // thread when a track with the same session ID is created 6059 if (io == 0 && mPlaybackThreads.size()) { 6060 io = mPlaybackThreads.keyAt(0); 6061 } 6062 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6063 } 6064 ThreadBase *thread = checkRecordThread_l(io); 6065 if (thread == NULL) { 6066 thread = checkPlaybackThread_l(io); 6067 if (thread == NULL) { 6068 ALOGE("createEffect() unknown output thread"); 6069 lStatus = BAD_VALUE; 6070 goto Exit; 6071 } 6072 } 6073 6074 sp<Client> client = registerPid_l(pid); 6075 6076 // create effect on selected output thread 6077 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6078 &desc, enabled, &lStatus); 6079 if (handle != 0 && id != NULL) { 6080 *id = handle->id(); 6081 } 6082 } 6083 6084Exit: 6085 if(status) { 6086 *status = lStatus; 6087 } 6088 return handle; 6089} 6090 6091status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6092 audio_io_handle_t dstOutput) 6093{ 6094 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6095 sessionId, srcOutput, dstOutput); 6096 Mutex::Autolock _l(mLock); 6097 if (srcOutput == dstOutput) { 6098 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6099 return NO_ERROR; 6100 } 6101 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6102 if (srcThread == NULL) { 6103 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6104 return BAD_VALUE; 6105 } 6106 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6107 if (dstThread == NULL) { 6108 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6109 return BAD_VALUE; 6110 } 6111 6112 Mutex::Autolock _dl(dstThread->mLock); 6113 Mutex::Autolock _sl(srcThread->mLock); 6114 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6115 6116 return NO_ERROR; 6117} 6118 6119// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6120status_t AudioFlinger::moveEffectChain_l(int sessionId, 6121 AudioFlinger::PlaybackThread *srcThread, 6122 AudioFlinger::PlaybackThread *dstThread, 6123 bool reRegister) 6124{ 6125 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6126 sessionId, srcThread, dstThread); 6127 6128 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6129 if (chain == 0) { 6130 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6131 sessionId, srcThread); 6132 return INVALID_OPERATION; 6133 } 6134 6135 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6136 // so that a new chain is created with correct parameters when first effect is added. This is 6137 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6138 // removed. 6139 srcThread->removeEffectChain_l(chain); 6140 6141 // transfer all effects one by one so that new effect chain is created on new thread with 6142 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6143 audio_io_handle_t dstOutput = dstThread->id(); 6144 sp<EffectChain> dstChain; 6145 uint32_t strategy = 0; // prevent compiler warning 6146 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6147 while (effect != 0) { 6148 srcThread->removeEffect_l(effect); 6149 dstThread->addEffect_l(effect); 6150 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6151 if (effect->state() == EffectModule::ACTIVE || 6152 effect->state() == EffectModule::STOPPING) { 6153 effect->start(); 6154 } 6155 // if the move request is not received from audio policy manager, the effect must be 6156 // re-registered with the new strategy and output 6157 if (dstChain == 0) { 6158 dstChain = effect->chain().promote(); 6159 if (dstChain == 0) { 6160 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6161 srcThread->addEffect_l(effect); 6162 return NO_INIT; 6163 } 6164 strategy = dstChain->strategy(); 6165 } 6166 if (reRegister) { 6167 AudioSystem::unregisterEffect(effect->id()); 6168 AudioSystem::registerEffect(&effect->desc(), 6169 dstOutput, 6170 strategy, 6171 sessionId, 6172 effect->id()); 6173 } 6174 effect = chain->getEffectFromId_l(0); 6175 } 6176 6177 return NO_ERROR; 6178} 6179 6180 6181// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6182sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6183 const sp<AudioFlinger::Client>& client, 6184 const sp<IEffectClient>& effectClient, 6185 int32_t priority, 6186 int sessionId, 6187 effect_descriptor_t *desc, 6188 int *enabled, 6189 status_t *status 6190 ) 6191{ 6192 sp<EffectModule> effect; 6193 sp<EffectHandle> handle; 6194 status_t lStatus; 6195 sp<EffectChain> chain; 6196 bool chainCreated = false; 6197 bool effectCreated = false; 6198 bool effectRegistered = false; 6199 6200 lStatus = initCheck(); 6201 if (lStatus != NO_ERROR) { 6202 ALOGW("createEffect_l() Audio driver not initialized."); 6203 goto Exit; 6204 } 6205 6206 // Do not allow effects with session ID 0 on direct output or duplicating threads 6207 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6208 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6209 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6210 desc->name, sessionId); 6211 lStatus = BAD_VALUE; 6212 goto Exit; 6213 } 6214 // Only Pre processor effects are allowed on input threads and only on input threads 6215 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6216 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6217 desc->name, desc->flags, mType); 6218 lStatus = BAD_VALUE; 6219 goto Exit; 6220 } 6221 6222 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6223 6224 { // scope for mLock 6225 Mutex::Autolock _l(mLock); 6226 6227 // check for existing effect chain with the requested audio session 6228 chain = getEffectChain_l(sessionId); 6229 if (chain == 0) { 6230 // create a new chain for this session 6231 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6232 chain = new EffectChain(this, sessionId); 6233 addEffectChain_l(chain); 6234 chain->setStrategy(getStrategyForSession_l(sessionId)); 6235 chainCreated = true; 6236 } else { 6237 effect = chain->getEffectFromDesc_l(desc); 6238 } 6239 6240 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6241 6242 if (effect == 0) { 6243 int id = mAudioFlinger->nextUniqueId(); 6244 // Check CPU and memory usage 6245 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6246 if (lStatus != NO_ERROR) { 6247 goto Exit; 6248 } 6249 effectRegistered = true; 6250 // create a new effect module if none present in the chain 6251 effect = new EffectModule(this, chain, desc, id, sessionId); 6252 lStatus = effect->status(); 6253 if (lStatus != NO_ERROR) { 6254 goto Exit; 6255 } 6256 lStatus = chain->addEffect_l(effect); 6257 if (lStatus != NO_ERROR) { 6258 goto Exit; 6259 } 6260 effectCreated = true; 6261 6262 effect->setDevice(mDevice); 6263 effect->setMode(mAudioFlinger->getMode()); 6264 } 6265 // create effect handle and connect it to effect module 6266 handle = new EffectHandle(effect, client, effectClient, priority); 6267 lStatus = effect->addHandle(handle); 6268 if (enabled != NULL) { 6269 *enabled = (int)effect->isEnabled(); 6270 } 6271 } 6272 6273Exit: 6274 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6275 Mutex::Autolock _l(mLock); 6276 if (effectCreated) { 6277 chain->removeEffect_l(effect); 6278 } 6279 if (effectRegistered) { 6280 AudioSystem::unregisterEffect(effect->id()); 6281 } 6282 if (chainCreated) { 6283 removeEffectChain_l(chain); 6284 } 6285 handle.clear(); 6286 } 6287 6288 if(status) { 6289 *status = lStatus; 6290 } 6291 return handle; 6292} 6293 6294sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6295{ 6296 sp<EffectChain> chain = getEffectChain_l(sessionId); 6297 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6298} 6299 6300// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6301// PlaybackThread::mLock held 6302status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6303{ 6304 // check for existing effect chain with the requested audio session 6305 int sessionId = effect->sessionId(); 6306 sp<EffectChain> chain = getEffectChain_l(sessionId); 6307 bool chainCreated = false; 6308 6309 if (chain == 0) { 6310 // create a new chain for this session 6311 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6312 chain = new EffectChain(this, sessionId); 6313 addEffectChain_l(chain); 6314 chain->setStrategy(getStrategyForSession_l(sessionId)); 6315 chainCreated = true; 6316 } 6317 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6318 6319 if (chain->getEffectFromId_l(effect->id()) != 0) { 6320 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6321 this, effect->desc().name, chain.get()); 6322 return BAD_VALUE; 6323 } 6324 6325 status_t status = chain->addEffect_l(effect); 6326 if (status != NO_ERROR) { 6327 if (chainCreated) { 6328 removeEffectChain_l(chain); 6329 } 6330 return status; 6331 } 6332 6333 effect->setDevice(mDevice); 6334 effect->setMode(mAudioFlinger->getMode()); 6335 return NO_ERROR; 6336} 6337 6338void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6339 6340 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6341 effect_descriptor_t desc = effect->desc(); 6342 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6343 detachAuxEffect_l(effect->id()); 6344 } 6345 6346 sp<EffectChain> chain = effect->chain().promote(); 6347 if (chain != 0) { 6348 // remove effect chain if removing last effect 6349 if (chain->removeEffect_l(effect) == 0) { 6350 removeEffectChain_l(chain); 6351 } 6352 } else { 6353 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6354 } 6355} 6356 6357void AudioFlinger::ThreadBase::lockEffectChains_l( 6358 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6359{ 6360 effectChains = mEffectChains; 6361 for (size_t i = 0; i < mEffectChains.size(); i++) { 6362 mEffectChains[i]->lock(); 6363 } 6364} 6365 6366void AudioFlinger::ThreadBase::unlockEffectChains( 6367 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6368{ 6369 for (size_t i = 0; i < effectChains.size(); i++) { 6370 effectChains[i]->unlock(); 6371 } 6372} 6373 6374sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6375{ 6376 Mutex::Autolock _l(mLock); 6377 return getEffectChain_l(sessionId); 6378} 6379 6380sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6381{ 6382 size_t size = mEffectChains.size(); 6383 for (size_t i = 0; i < size; i++) { 6384 if (mEffectChains[i]->sessionId() == sessionId) { 6385 return mEffectChains[i]; 6386 } 6387 } 6388 return 0; 6389} 6390 6391void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6392{ 6393 Mutex::Autolock _l(mLock); 6394 size_t size = mEffectChains.size(); 6395 for (size_t i = 0; i < size; i++) { 6396 mEffectChains[i]->setMode_l(mode); 6397 } 6398} 6399 6400void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6401 const wp<EffectHandle>& handle, 6402 bool unpinIfLast) { 6403 6404 Mutex::Autolock _l(mLock); 6405 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6406 // delete the effect module if removing last handle on it 6407 if (effect->removeHandle(handle) == 0) { 6408 if (!effect->isPinned() || unpinIfLast) { 6409 removeEffect_l(effect); 6410 AudioSystem::unregisterEffect(effect->id()); 6411 } 6412 } 6413} 6414 6415status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6416{ 6417 int session = chain->sessionId(); 6418 int16_t *buffer = mMixBuffer; 6419 bool ownsBuffer = false; 6420 6421 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6422 if (session > 0) { 6423 // Only one effect chain can be present in direct output thread and it uses 6424 // the mix buffer as input 6425 if (mType != DIRECT) { 6426 size_t numSamples = mFrameCount * mChannelCount; 6427 buffer = new int16_t[numSamples]; 6428 memset(buffer, 0, numSamples * sizeof(int16_t)); 6429 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6430 ownsBuffer = true; 6431 } 6432 6433 // Attach all tracks with same session ID to this chain. 6434 for (size_t i = 0; i < mTracks.size(); ++i) { 6435 sp<Track> track = mTracks[i]; 6436 if (session == track->sessionId()) { 6437 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6438 track->setMainBuffer(buffer); 6439 chain->incTrackCnt(); 6440 } 6441 } 6442 6443 // indicate all active tracks in the chain 6444 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6445 sp<Track> track = mActiveTracks[i].promote(); 6446 if (track == 0) continue; 6447 if (session == track->sessionId()) { 6448 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6449 chain->incActiveTrackCnt(); 6450 } 6451 } 6452 } 6453 6454 chain->setInBuffer(buffer, ownsBuffer); 6455 chain->setOutBuffer(mMixBuffer); 6456 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6457 // chains list in order to be processed last as it contains output stage effects 6458 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6459 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6460 // after track specific effects and before output stage 6461 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6462 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6463 // Effect chain for other sessions are inserted at beginning of effect 6464 // chains list to be processed before output mix effects. Relative order between other 6465 // sessions is not important 6466 size_t size = mEffectChains.size(); 6467 size_t i = 0; 6468 for (i = 0; i < size; i++) { 6469 if (mEffectChains[i]->sessionId() < session) break; 6470 } 6471 mEffectChains.insertAt(chain, i); 6472 checkSuspendOnAddEffectChain_l(chain); 6473 6474 return NO_ERROR; 6475} 6476 6477size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6478{ 6479 int session = chain->sessionId(); 6480 6481 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6482 6483 for (size_t i = 0; i < mEffectChains.size(); i++) { 6484 if (chain == mEffectChains[i]) { 6485 mEffectChains.removeAt(i); 6486 // detach all active tracks from the chain 6487 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6488 sp<Track> track = mActiveTracks[i].promote(); 6489 if (track == 0) continue; 6490 if (session == track->sessionId()) { 6491 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6492 chain.get(), session); 6493 chain->decActiveTrackCnt(); 6494 } 6495 } 6496 6497 // detach all tracks with same session ID from this chain 6498 for (size_t i = 0; i < mTracks.size(); ++i) { 6499 sp<Track> track = mTracks[i]; 6500 if (session == track->sessionId()) { 6501 track->setMainBuffer(mMixBuffer); 6502 chain->decTrackCnt(); 6503 } 6504 } 6505 break; 6506 } 6507 } 6508 return mEffectChains.size(); 6509} 6510 6511status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6512 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6513{ 6514 Mutex::Autolock _l(mLock); 6515 return attachAuxEffect_l(track, EffectId); 6516} 6517 6518status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6519 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6520{ 6521 status_t status = NO_ERROR; 6522 6523 if (EffectId == 0) { 6524 track->setAuxBuffer(0, NULL); 6525 } else { 6526 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6527 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6528 if (effect != 0) { 6529 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6530 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6531 } else { 6532 status = INVALID_OPERATION; 6533 } 6534 } else { 6535 status = BAD_VALUE; 6536 } 6537 } 6538 return status; 6539} 6540 6541void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6542{ 6543 for (size_t i = 0; i < mTracks.size(); ++i) { 6544 sp<Track> track = mTracks[i]; 6545 if (track->auxEffectId() == effectId) { 6546 attachAuxEffect_l(track, 0); 6547 } 6548 } 6549} 6550 6551status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6552{ 6553 // only one chain per input thread 6554 if (mEffectChains.size() != 0) { 6555 return INVALID_OPERATION; 6556 } 6557 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6558 6559 chain->setInBuffer(NULL); 6560 chain->setOutBuffer(NULL); 6561 6562 checkSuspendOnAddEffectChain_l(chain); 6563 6564 mEffectChains.add(chain); 6565 6566 return NO_ERROR; 6567} 6568 6569size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6570{ 6571 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6572 ALOGW_IF(mEffectChains.size() != 1, 6573 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6574 chain.get(), mEffectChains.size(), this); 6575 if (mEffectChains.size() == 1) { 6576 mEffectChains.removeAt(0); 6577 } 6578 return 0; 6579} 6580 6581// ---------------------------------------------------------------------------- 6582// EffectModule implementation 6583// ---------------------------------------------------------------------------- 6584 6585#undef LOG_TAG 6586#define LOG_TAG "AudioFlinger::EffectModule" 6587 6588AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6589 const wp<AudioFlinger::EffectChain>& chain, 6590 effect_descriptor_t *desc, 6591 int id, 6592 int sessionId) 6593 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6594 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6595{ 6596 ALOGV("Constructor %p", this); 6597 int lStatus; 6598 if (thread == NULL) { 6599 return; 6600 } 6601 6602 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6603 6604 // create effect engine from effect factory 6605 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6606 6607 if (mStatus != NO_ERROR) { 6608 return; 6609 } 6610 lStatus = init(); 6611 if (lStatus < 0) { 6612 mStatus = lStatus; 6613 goto Error; 6614 } 6615 6616 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6617 mPinned = true; 6618 } 6619 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6620 return; 6621Error: 6622 EffectRelease(mEffectInterface); 6623 mEffectInterface = NULL; 6624 ALOGV("Constructor Error %d", mStatus); 6625} 6626 6627AudioFlinger::EffectModule::~EffectModule() 6628{ 6629 ALOGV("Destructor %p", this); 6630 if (mEffectInterface != NULL) { 6631 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6632 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6633 sp<ThreadBase> thread = mThread.promote(); 6634 if (thread != 0) { 6635 audio_stream_t *stream = thread->stream(); 6636 if (stream != NULL) { 6637 stream->remove_audio_effect(stream, mEffectInterface); 6638 } 6639 } 6640 } 6641 // release effect engine 6642 EffectRelease(mEffectInterface); 6643 } 6644} 6645 6646status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6647{ 6648 status_t status; 6649 6650 Mutex::Autolock _l(mLock); 6651 int priority = handle->priority(); 6652 size_t size = mHandles.size(); 6653 sp<EffectHandle> h; 6654 size_t i; 6655 for (i = 0; i < size; i++) { 6656 h = mHandles[i].promote(); 6657 if (h == 0) continue; 6658 if (h->priority() <= priority) break; 6659 } 6660 // if inserted in first place, move effect control from previous owner to this handle 6661 if (i == 0) { 6662 bool enabled = false; 6663 if (h != 0) { 6664 enabled = h->enabled(); 6665 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6666 } 6667 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6668 status = NO_ERROR; 6669 } else { 6670 status = ALREADY_EXISTS; 6671 } 6672 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6673 mHandles.insertAt(handle, i); 6674 return status; 6675} 6676 6677size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6678{ 6679 Mutex::Autolock _l(mLock); 6680 size_t size = mHandles.size(); 6681 size_t i; 6682 for (i = 0; i < size; i++) { 6683 if (mHandles[i] == handle) break; 6684 } 6685 if (i == size) { 6686 return size; 6687 } 6688 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6689 6690 bool enabled = false; 6691 EffectHandle *hdl = handle.unsafe_get(); 6692 if (hdl != NULL) { 6693 ALOGV("removeHandle() unsafe_get OK"); 6694 enabled = hdl->enabled(); 6695 } 6696 mHandles.removeAt(i); 6697 size = mHandles.size(); 6698 // if removed from first place, move effect control from this handle to next in line 6699 if (i == 0 && size != 0) { 6700 sp<EffectHandle> h = mHandles[0].promote(); 6701 if (h != 0) { 6702 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6703 } 6704 } 6705 6706 // Prevent calls to process() and other functions on effect interface from now on. 6707 // The effect engine will be released by the destructor when the last strong reference on 6708 // this object is released which can happen after next process is called. 6709 if (size == 0 && !mPinned) { 6710 mState = DESTROYED; 6711 } 6712 6713 return size; 6714} 6715 6716sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6717{ 6718 Mutex::Autolock _l(mLock); 6719 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6720} 6721 6722void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6723{ 6724 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6725 // keep a strong reference on this EffectModule to avoid calling the 6726 // destructor before we exit 6727 sp<EffectModule> keep(this); 6728 { 6729 sp<ThreadBase> thread = mThread.promote(); 6730 if (thread != 0) { 6731 thread->disconnectEffect(keep, handle, unpinIfLast); 6732 } 6733 } 6734} 6735 6736void AudioFlinger::EffectModule::updateState() { 6737 Mutex::Autolock _l(mLock); 6738 6739 switch (mState) { 6740 case RESTART: 6741 reset_l(); 6742 // FALL THROUGH 6743 6744 case STARTING: 6745 // clear auxiliary effect input buffer for next accumulation 6746 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6747 memset(mConfig.inputCfg.buffer.raw, 6748 0, 6749 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6750 } 6751 start_l(); 6752 mState = ACTIVE; 6753 break; 6754 case STOPPING: 6755 stop_l(); 6756 mDisableWaitCnt = mMaxDisableWaitCnt; 6757 mState = STOPPED; 6758 break; 6759 case STOPPED: 6760 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6761 // turn off sequence. 6762 if (--mDisableWaitCnt == 0) { 6763 reset_l(); 6764 mState = IDLE; 6765 } 6766 break; 6767 default: //IDLE , ACTIVE, DESTROYED 6768 break; 6769 } 6770} 6771 6772void AudioFlinger::EffectModule::process() 6773{ 6774 Mutex::Autolock _l(mLock); 6775 6776 if (mState == DESTROYED || mEffectInterface == NULL || 6777 mConfig.inputCfg.buffer.raw == NULL || 6778 mConfig.outputCfg.buffer.raw == NULL) { 6779 return; 6780 } 6781 6782 if (isProcessEnabled()) { 6783 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6784 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6785 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6786 mConfig.inputCfg.buffer.s32, 6787 mConfig.inputCfg.buffer.frameCount/2); 6788 } 6789 6790 // do the actual processing in the effect engine 6791 int ret = (*mEffectInterface)->process(mEffectInterface, 6792 &mConfig.inputCfg.buffer, 6793 &mConfig.outputCfg.buffer); 6794 6795 // force transition to IDLE state when engine is ready 6796 if (mState == STOPPED && ret == -ENODATA) { 6797 mDisableWaitCnt = 1; 6798 } 6799 6800 // clear auxiliary effect input buffer for next accumulation 6801 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6802 memset(mConfig.inputCfg.buffer.raw, 0, 6803 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6804 } 6805 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6806 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6807 // If an insert effect is idle and input buffer is different from output buffer, 6808 // accumulate input onto output 6809 sp<EffectChain> chain = mChain.promote(); 6810 if (chain != 0 && chain->activeTrackCnt() != 0) { 6811 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6812 int16_t *in = mConfig.inputCfg.buffer.s16; 6813 int16_t *out = mConfig.outputCfg.buffer.s16; 6814 for (size_t i = 0; i < frameCnt; i++) { 6815 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6816 } 6817 } 6818 } 6819} 6820 6821void AudioFlinger::EffectModule::reset_l() 6822{ 6823 if (mEffectInterface == NULL) { 6824 return; 6825 } 6826 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6827} 6828 6829status_t AudioFlinger::EffectModule::configure() 6830{ 6831 uint32_t channels; 6832 if (mEffectInterface == NULL) { 6833 return NO_INIT; 6834 } 6835 6836 sp<ThreadBase> thread = mThread.promote(); 6837 if (thread == 0) { 6838 return DEAD_OBJECT; 6839 } 6840 6841 // TODO: handle configuration of effects replacing track process 6842 if (thread->channelCount() == 1) { 6843 channels = AUDIO_CHANNEL_OUT_MONO; 6844 } else { 6845 channels = AUDIO_CHANNEL_OUT_STEREO; 6846 } 6847 6848 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6849 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6850 } else { 6851 mConfig.inputCfg.channels = channels; 6852 } 6853 mConfig.outputCfg.channels = channels; 6854 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6855 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6856 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6857 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6858 mConfig.inputCfg.bufferProvider.cookie = NULL; 6859 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6860 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6861 mConfig.outputCfg.bufferProvider.cookie = NULL; 6862 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6863 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6864 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6865 // Insert effect: 6866 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6867 // always overwrites output buffer: input buffer == output buffer 6868 // - in other sessions: 6869 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6870 // other effect: overwrites output buffer: input buffer == output buffer 6871 // Auxiliary effect: 6872 // accumulates in output buffer: input buffer != output buffer 6873 // Therefore: accumulate <=> input buffer != output buffer 6874 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6875 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6876 } else { 6877 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6878 } 6879 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6880 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6881 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6882 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6883 6884 ALOGV("configure() %p thread %p buffer %p framecount %d", 6885 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6886 6887 status_t cmdStatus; 6888 uint32_t size = sizeof(int); 6889 status_t status = (*mEffectInterface)->command(mEffectInterface, 6890 EFFECT_CMD_SET_CONFIG, 6891 sizeof(effect_config_t), 6892 &mConfig, 6893 &size, 6894 &cmdStatus); 6895 if (status == 0) { 6896 status = cmdStatus; 6897 } 6898 6899 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6900 (1000 * mConfig.outputCfg.buffer.frameCount); 6901 6902 return status; 6903} 6904 6905status_t AudioFlinger::EffectModule::init() 6906{ 6907 Mutex::Autolock _l(mLock); 6908 if (mEffectInterface == NULL) { 6909 return NO_INIT; 6910 } 6911 status_t cmdStatus; 6912 uint32_t size = sizeof(status_t); 6913 status_t status = (*mEffectInterface)->command(mEffectInterface, 6914 EFFECT_CMD_INIT, 6915 0, 6916 NULL, 6917 &size, 6918 &cmdStatus); 6919 if (status == 0) { 6920 status = cmdStatus; 6921 } 6922 return status; 6923} 6924 6925status_t AudioFlinger::EffectModule::start() 6926{ 6927 Mutex::Autolock _l(mLock); 6928 return start_l(); 6929} 6930 6931status_t AudioFlinger::EffectModule::start_l() 6932{ 6933 if (mEffectInterface == NULL) { 6934 return NO_INIT; 6935 } 6936 status_t cmdStatus; 6937 uint32_t size = sizeof(status_t); 6938 status_t status = (*mEffectInterface)->command(mEffectInterface, 6939 EFFECT_CMD_ENABLE, 6940 0, 6941 NULL, 6942 &size, 6943 &cmdStatus); 6944 if (status == 0) { 6945 status = cmdStatus; 6946 } 6947 if (status == 0 && 6948 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6949 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6950 sp<ThreadBase> thread = mThread.promote(); 6951 if (thread != 0) { 6952 audio_stream_t *stream = thread->stream(); 6953 if (stream != NULL) { 6954 stream->add_audio_effect(stream, mEffectInterface); 6955 } 6956 } 6957 } 6958 return status; 6959} 6960 6961status_t AudioFlinger::EffectModule::stop() 6962{ 6963 Mutex::Autolock _l(mLock); 6964 return stop_l(); 6965} 6966 6967status_t AudioFlinger::EffectModule::stop_l() 6968{ 6969 if (mEffectInterface == NULL) { 6970 return NO_INIT; 6971 } 6972 status_t cmdStatus; 6973 uint32_t size = sizeof(status_t); 6974 status_t status = (*mEffectInterface)->command(mEffectInterface, 6975 EFFECT_CMD_DISABLE, 6976 0, 6977 NULL, 6978 &size, 6979 &cmdStatus); 6980 if (status == 0) { 6981 status = cmdStatus; 6982 } 6983 if (status == 0 && 6984 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6985 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6986 sp<ThreadBase> thread = mThread.promote(); 6987 if (thread != 0) { 6988 audio_stream_t *stream = thread->stream(); 6989 if (stream != NULL) { 6990 stream->remove_audio_effect(stream, mEffectInterface); 6991 } 6992 } 6993 } 6994 return status; 6995} 6996 6997status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6998 uint32_t cmdSize, 6999 void *pCmdData, 7000 uint32_t *replySize, 7001 void *pReplyData) 7002{ 7003 Mutex::Autolock _l(mLock); 7004// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7005 7006 if (mState == DESTROYED || mEffectInterface == NULL) { 7007 return NO_INIT; 7008 } 7009 status_t status = (*mEffectInterface)->command(mEffectInterface, 7010 cmdCode, 7011 cmdSize, 7012 pCmdData, 7013 replySize, 7014 pReplyData); 7015 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7016 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7017 for (size_t i = 1; i < mHandles.size(); i++) { 7018 sp<EffectHandle> h = mHandles[i].promote(); 7019 if (h != 0) { 7020 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7021 } 7022 } 7023 } 7024 return status; 7025} 7026 7027status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7028{ 7029 7030 Mutex::Autolock _l(mLock); 7031 ALOGV("setEnabled %p enabled %d", this, enabled); 7032 7033 if (enabled != isEnabled()) { 7034 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7035 if (enabled && status != NO_ERROR) { 7036 return status; 7037 } 7038 7039 switch (mState) { 7040 // going from disabled to enabled 7041 case IDLE: 7042 mState = STARTING; 7043 break; 7044 case STOPPED: 7045 mState = RESTART; 7046 break; 7047 case STOPPING: 7048 mState = ACTIVE; 7049 break; 7050 7051 // going from enabled to disabled 7052 case RESTART: 7053 mState = STOPPED; 7054 break; 7055 case STARTING: 7056 mState = IDLE; 7057 break; 7058 case ACTIVE: 7059 mState = STOPPING; 7060 break; 7061 case DESTROYED: 7062 return NO_ERROR; // simply ignore as we are being destroyed 7063 } 7064 for (size_t i = 1; i < mHandles.size(); i++) { 7065 sp<EffectHandle> h = mHandles[i].promote(); 7066 if (h != 0) { 7067 h->setEnabled(enabled); 7068 } 7069 } 7070 } 7071 return NO_ERROR; 7072} 7073 7074bool AudioFlinger::EffectModule::isEnabled() const 7075{ 7076 switch (mState) { 7077 case RESTART: 7078 case STARTING: 7079 case ACTIVE: 7080 return true; 7081 case IDLE: 7082 case STOPPING: 7083 case STOPPED: 7084 case DESTROYED: 7085 default: 7086 return false; 7087 } 7088} 7089 7090bool AudioFlinger::EffectModule::isProcessEnabled() const 7091{ 7092 switch (mState) { 7093 case RESTART: 7094 case ACTIVE: 7095 case STOPPING: 7096 case STOPPED: 7097 return true; 7098 case IDLE: 7099 case STARTING: 7100 case DESTROYED: 7101 default: 7102 return false; 7103 } 7104} 7105 7106status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7107{ 7108 Mutex::Autolock _l(mLock); 7109 status_t status = NO_ERROR; 7110 7111 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7112 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7113 if (isProcessEnabled() && 7114 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7115 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7116 status_t cmdStatus; 7117 uint32_t volume[2]; 7118 uint32_t *pVolume = NULL; 7119 uint32_t size = sizeof(volume); 7120 volume[0] = *left; 7121 volume[1] = *right; 7122 if (controller) { 7123 pVolume = volume; 7124 } 7125 status = (*mEffectInterface)->command(mEffectInterface, 7126 EFFECT_CMD_SET_VOLUME, 7127 size, 7128 volume, 7129 &size, 7130 pVolume); 7131 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7132 *left = volume[0]; 7133 *right = volume[1]; 7134 } 7135 } 7136 return status; 7137} 7138 7139status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7140{ 7141 Mutex::Autolock _l(mLock); 7142 status_t status = NO_ERROR; 7143 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7144 // audio pre processing modules on RecordThread can receive both output and 7145 // input device indication in the same call 7146 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7147 if (dev) { 7148 status_t cmdStatus; 7149 uint32_t size = sizeof(status_t); 7150 7151 status = (*mEffectInterface)->command(mEffectInterface, 7152 EFFECT_CMD_SET_DEVICE, 7153 sizeof(uint32_t), 7154 &dev, 7155 &size, 7156 &cmdStatus); 7157 if (status == NO_ERROR) { 7158 status = cmdStatus; 7159 } 7160 } 7161 dev = device & AUDIO_DEVICE_IN_ALL; 7162 if (dev) { 7163 status_t cmdStatus; 7164 uint32_t size = sizeof(status_t); 7165 7166 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7167 EFFECT_CMD_SET_INPUT_DEVICE, 7168 sizeof(uint32_t), 7169 &dev, 7170 &size, 7171 &cmdStatus); 7172 if (status2 == NO_ERROR) { 7173 status2 = cmdStatus; 7174 } 7175 if (status == NO_ERROR) { 7176 status = status2; 7177 } 7178 } 7179 } 7180 return status; 7181} 7182 7183status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7184{ 7185 Mutex::Autolock _l(mLock); 7186 status_t status = NO_ERROR; 7187 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7188 status_t cmdStatus; 7189 uint32_t size = sizeof(status_t); 7190 status = (*mEffectInterface)->command(mEffectInterface, 7191 EFFECT_CMD_SET_AUDIO_MODE, 7192 sizeof(audio_mode_t), 7193 &mode, 7194 &size, 7195 &cmdStatus); 7196 if (status == NO_ERROR) { 7197 status = cmdStatus; 7198 } 7199 } 7200 return status; 7201} 7202 7203void AudioFlinger::EffectModule::setSuspended(bool suspended) 7204{ 7205 Mutex::Autolock _l(mLock); 7206 mSuspended = suspended; 7207} 7208 7209bool AudioFlinger::EffectModule::suspended() const 7210{ 7211 Mutex::Autolock _l(mLock); 7212 return mSuspended; 7213} 7214 7215status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7216{ 7217 const size_t SIZE = 256; 7218 char buffer[SIZE]; 7219 String8 result; 7220 7221 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7222 result.append(buffer); 7223 7224 bool locked = tryLock(mLock); 7225 // failed to lock - AudioFlinger is probably deadlocked 7226 if (!locked) { 7227 result.append("\t\tCould not lock Fx mutex:\n"); 7228 } 7229 7230 result.append("\t\tSession Status State Engine:\n"); 7231 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7232 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7233 result.append(buffer); 7234 7235 result.append("\t\tDescriptor:\n"); 7236 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7237 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7238 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7239 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7240 result.append(buffer); 7241 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7242 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7243 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7244 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7245 result.append(buffer); 7246 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7247 mDescriptor.apiVersion, 7248 mDescriptor.flags); 7249 result.append(buffer); 7250 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7251 mDescriptor.name); 7252 result.append(buffer); 7253 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7254 mDescriptor.implementor); 7255 result.append(buffer); 7256 7257 result.append("\t\t- Input configuration:\n"); 7258 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7259 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7260 (uint32_t)mConfig.inputCfg.buffer.raw, 7261 mConfig.inputCfg.buffer.frameCount, 7262 mConfig.inputCfg.samplingRate, 7263 mConfig.inputCfg.channels, 7264 mConfig.inputCfg.format); 7265 result.append(buffer); 7266 7267 result.append("\t\t- Output configuration:\n"); 7268 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7269 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7270 (uint32_t)mConfig.outputCfg.buffer.raw, 7271 mConfig.outputCfg.buffer.frameCount, 7272 mConfig.outputCfg.samplingRate, 7273 mConfig.outputCfg.channels, 7274 mConfig.outputCfg.format); 7275 result.append(buffer); 7276 7277 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7278 result.append(buffer); 7279 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7280 for (size_t i = 0; i < mHandles.size(); ++i) { 7281 sp<EffectHandle> handle = mHandles[i].promote(); 7282 if (handle != 0) { 7283 handle->dump(buffer, SIZE); 7284 result.append(buffer); 7285 } 7286 } 7287 7288 result.append("\n"); 7289 7290 write(fd, result.string(), result.length()); 7291 7292 if (locked) { 7293 mLock.unlock(); 7294 } 7295 7296 return NO_ERROR; 7297} 7298 7299// ---------------------------------------------------------------------------- 7300// EffectHandle implementation 7301// ---------------------------------------------------------------------------- 7302 7303#undef LOG_TAG 7304#define LOG_TAG "AudioFlinger::EffectHandle" 7305 7306AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7307 const sp<AudioFlinger::Client>& client, 7308 const sp<IEffectClient>& effectClient, 7309 int32_t priority) 7310 : BnEffect(), 7311 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7312 mPriority(priority), mHasControl(false), mEnabled(false) 7313{ 7314 ALOGV("constructor %p", this); 7315 7316 if (client == 0) { 7317 return; 7318 } 7319 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7320 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7321 if (mCblkMemory != 0) { 7322 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7323 7324 if (mCblk != NULL) { 7325 new(mCblk) effect_param_cblk_t(); 7326 mBuffer = (uint8_t *)mCblk + bufOffset; 7327 } 7328 } else { 7329 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7330 return; 7331 } 7332} 7333 7334AudioFlinger::EffectHandle::~EffectHandle() 7335{ 7336 ALOGV("Destructor %p", this); 7337 disconnect(false); 7338 ALOGV("Destructor DONE %p", this); 7339} 7340 7341status_t AudioFlinger::EffectHandle::enable() 7342{ 7343 ALOGV("enable %p", this); 7344 if (!mHasControl) return INVALID_OPERATION; 7345 if (mEffect == 0) return DEAD_OBJECT; 7346 7347 if (mEnabled) { 7348 return NO_ERROR; 7349 } 7350 7351 mEnabled = true; 7352 7353 sp<ThreadBase> thread = mEffect->thread().promote(); 7354 if (thread != 0) { 7355 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7356 } 7357 7358 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7359 if (mEffect->suspended()) { 7360 return NO_ERROR; 7361 } 7362 7363 status_t status = mEffect->setEnabled(true); 7364 if (status != NO_ERROR) { 7365 if (thread != 0) { 7366 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7367 } 7368 mEnabled = false; 7369 } 7370 return status; 7371} 7372 7373status_t AudioFlinger::EffectHandle::disable() 7374{ 7375 ALOGV("disable %p", this); 7376 if (!mHasControl) return INVALID_OPERATION; 7377 if (mEffect == 0) return DEAD_OBJECT; 7378 7379 if (!mEnabled) { 7380 return NO_ERROR; 7381 } 7382 mEnabled = false; 7383 7384 if (mEffect->suspended()) { 7385 return NO_ERROR; 7386 } 7387 7388 status_t status = mEffect->setEnabled(false); 7389 7390 sp<ThreadBase> thread = mEffect->thread().promote(); 7391 if (thread != 0) { 7392 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7393 } 7394 7395 return status; 7396} 7397 7398void AudioFlinger::EffectHandle::disconnect() 7399{ 7400 disconnect(true); 7401} 7402 7403void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7404{ 7405 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7406 if (mEffect == 0) { 7407 return; 7408 } 7409 mEffect->disconnect(this, unpinIfLast); 7410 7411 if (mHasControl && mEnabled) { 7412 sp<ThreadBase> thread = mEffect->thread().promote(); 7413 if (thread != 0) { 7414 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7415 } 7416 } 7417 7418 // release sp on module => module destructor can be called now 7419 mEffect.clear(); 7420 if (mClient != 0) { 7421 if (mCblk != NULL) { 7422 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7423 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7424 } 7425 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7426 // Client destructor must run with AudioFlinger mutex locked 7427 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7428 mClient.clear(); 7429 } 7430} 7431 7432status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7433 uint32_t cmdSize, 7434 void *pCmdData, 7435 uint32_t *replySize, 7436 void *pReplyData) 7437{ 7438// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7439// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7440 7441 // only get parameter command is permitted for applications not controlling the effect 7442 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7443 return INVALID_OPERATION; 7444 } 7445 if (mEffect == 0) return DEAD_OBJECT; 7446 if (mClient == 0) return INVALID_OPERATION; 7447 7448 // handle commands that are not forwarded transparently to effect engine 7449 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7450 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7451 // no risk to block the whole media server process or mixer threads is we are stuck here 7452 Mutex::Autolock _l(mCblk->lock); 7453 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7454 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7455 mCblk->serverIndex = 0; 7456 mCblk->clientIndex = 0; 7457 return BAD_VALUE; 7458 } 7459 status_t status = NO_ERROR; 7460 while (mCblk->serverIndex < mCblk->clientIndex) { 7461 int reply; 7462 uint32_t rsize = sizeof(int); 7463 int *p = (int *)(mBuffer + mCblk->serverIndex); 7464 int size = *p++; 7465 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7466 ALOGW("command(): invalid parameter block size"); 7467 break; 7468 } 7469 effect_param_t *param = (effect_param_t *)p; 7470 if (param->psize == 0 || param->vsize == 0) { 7471 ALOGW("command(): null parameter or value size"); 7472 mCblk->serverIndex += size; 7473 continue; 7474 } 7475 uint32_t psize = sizeof(effect_param_t) + 7476 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7477 param->vsize; 7478 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7479 psize, 7480 p, 7481 &rsize, 7482 &reply); 7483 // stop at first error encountered 7484 if (ret != NO_ERROR) { 7485 status = ret; 7486 *(int *)pReplyData = reply; 7487 break; 7488 } else if (reply != NO_ERROR) { 7489 *(int *)pReplyData = reply; 7490 break; 7491 } 7492 mCblk->serverIndex += size; 7493 } 7494 mCblk->serverIndex = 0; 7495 mCblk->clientIndex = 0; 7496 return status; 7497 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7498 *(int *)pReplyData = NO_ERROR; 7499 return enable(); 7500 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7501 *(int *)pReplyData = NO_ERROR; 7502 return disable(); 7503 } 7504 7505 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7506} 7507 7508void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7509{ 7510 ALOGV("setControl %p control %d", this, hasControl); 7511 7512 mHasControl = hasControl; 7513 mEnabled = enabled; 7514 7515 if (signal && mEffectClient != 0) { 7516 mEffectClient->controlStatusChanged(hasControl); 7517 } 7518} 7519 7520void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7521 uint32_t cmdSize, 7522 void *pCmdData, 7523 uint32_t replySize, 7524 void *pReplyData) 7525{ 7526 if (mEffectClient != 0) { 7527 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7528 } 7529} 7530 7531 7532 7533void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7534{ 7535 if (mEffectClient != 0) { 7536 mEffectClient->enableStatusChanged(enabled); 7537 } 7538} 7539 7540status_t AudioFlinger::EffectHandle::onTransact( 7541 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7542{ 7543 return BnEffect::onTransact(code, data, reply, flags); 7544} 7545 7546 7547void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7548{ 7549 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7550 7551 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7552 (mClient == 0) ? getpid_cached : mClient->pid(), 7553 mPriority, 7554 mHasControl, 7555 !locked, 7556 mCblk ? mCblk->clientIndex : 0, 7557 mCblk ? mCblk->serverIndex : 0 7558 ); 7559 7560 if (locked) { 7561 mCblk->lock.unlock(); 7562 } 7563} 7564 7565#undef LOG_TAG 7566#define LOG_TAG "AudioFlinger::EffectChain" 7567 7568AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7569 int sessionId) 7570 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7571 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7572 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7573{ 7574 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7575 if (thread == NULL) { 7576 return; 7577 } 7578 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7579 thread->frameCount(); 7580} 7581 7582AudioFlinger::EffectChain::~EffectChain() 7583{ 7584 if (mOwnInBuffer) { 7585 delete mInBuffer; 7586 } 7587 7588} 7589 7590// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7591sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7592{ 7593 size_t size = mEffects.size(); 7594 7595 for (size_t i = 0; i < size; i++) { 7596 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7597 return mEffects[i]; 7598 } 7599 } 7600 return 0; 7601} 7602 7603// getEffectFromId_l() must be called with ThreadBase::mLock held 7604sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7605{ 7606 size_t size = mEffects.size(); 7607 7608 for (size_t i = 0; i < size; i++) { 7609 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7610 if (id == 0 || mEffects[i]->id() == id) { 7611 return mEffects[i]; 7612 } 7613 } 7614 return 0; 7615} 7616 7617// getEffectFromType_l() must be called with ThreadBase::mLock held 7618sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7619 const effect_uuid_t *type) 7620{ 7621 size_t size = mEffects.size(); 7622 7623 for (size_t i = 0; i < size; i++) { 7624 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7625 return mEffects[i]; 7626 } 7627 } 7628 return 0; 7629} 7630 7631// Must be called with EffectChain::mLock locked 7632void AudioFlinger::EffectChain::process_l() 7633{ 7634 sp<ThreadBase> thread = mThread.promote(); 7635 if (thread == 0) { 7636 ALOGW("process_l(): cannot promote mixer thread"); 7637 return; 7638 } 7639 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7640 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7641 // always process effects unless no more tracks are on the session and the effect tail 7642 // has been rendered 7643 bool doProcess = true; 7644 if (!isGlobalSession) { 7645 bool tracksOnSession = (trackCnt() != 0); 7646 7647 if (!tracksOnSession && mTailBufferCount == 0) { 7648 doProcess = false; 7649 } 7650 7651 if (activeTrackCnt() == 0) { 7652 // if no track is active and the effect tail has not been rendered, 7653 // the input buffer must be cleared here as the mixer process will not do it 7654 if (tracksOnSession || mTailBufferCount > 0) { 7655 size_t numSamples = thread->frameCount() * thread->channelCount(); 7656 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7657 if (mTailBufferCount > 0) { 7658 mTailBufferCount--; 7659 } 7660 } 7661 } 7662 } 7663 7664 size_t size = mEffects.size(); 7665 if (doProcess) { 7666 for (size_t i = 0; i < size; i++) { 7667 mEffects[i]->process(); 7668 } 7669 } 7670 for (size_t i = 0; i < size; i++) { 7671 mEffects[i]->updateState(); 7672 } 7673} 7674 7675// addEffect_l() must be called with PlaybackThread::mLock held 7676status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7677{ 7678 effect_descriptor_t desc = effect->desc(); 7679 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7680 7681 Mutex::Autolock _l(mLock); 7682 effect->setChain(this); 7683 sp<ThreadBase> thread = mThread.promote(); 7684 if (thread == 0) { 7685 return NO_INIT; 7686 } 7687 effect->setThread(thread); 7688 7689 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7690 // Auxiliary effects are inserted at the beginning of mEffects vector as 7691 // they are processed first and accumulated in chain input buffer 7692 mEffects.insertAt(effect, 0); 7693 7694 // the input buffer for auxiliary effect contains mono samples in 7695 // 32 bit format. This is to avoid saturation in AudoMixer 7696 // accumulation stage. Saturation is done in EffectModule::process() before 7697 // calling the process in effect engine 7698 size_t numSamples = thread->frameCount(); 7699 int32_t *buffer = new int32_t[numSamples]; 7700 memset(buffer, 0, numSamples * sizeof(int32_t)); 7701 effect->setInBuffer((int16_t *)buffer); 7702 // auxiliary effects output samples to chain input buffer for further processing 7703 // by insert effects 7704 effect->setOutBuffer(mInBuffer); 7705 } else { 7706 // Insert effects are inserted at the end of mEffects vector as they are processed 7707 // after track and auxiliary effects. 7708 // Insert effect order as a function of indicated preference: 7709 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7710 // another effect is present 7711 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7712 // last effect claiming first position 7713 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7714 // first effect claiming last position 7715 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7716 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7717 // already present 7718 7719 size_t size = mEffects.size(); 7720 size_t idx_insert = size; 7721 ssize_t idx_insert_first = -1; 7722 ssize_t idx_insert_last = -1; 7723 7724 for (size_t i = 0; i < size; i++) { 7725 effect_descriptor_t d = mEffects[i]->desc(); 7726 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7727 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7728 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7729 // check invalid effect chaining combinations 7730 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7731 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7732 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7733 return INVALID_OPERATION; 7734 } 7735 // remember position of first insert effect and by default 7736 // select this as insert position for new effect 7737 if (idx_insert == size) { 7738 idx_insert = i; 7739 } 7740 // remember position of last insert effect claiming 7741 // first position 7742 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7743 idx_insert_first = i; 7744 } 7745 // remember position of first insert effect claiming 7746 // last position 7747 if (iPref == EFFECT_FLAG_INSERT_LAST && 7748 idx_insert_last == -1) { 7749 idx_insert_last = i; 7750 } 7751 } 7752 } 7753 7754 // modify idx_insert from first position if needed 7755 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7756 if (idx_insert_last != -1) { 7757 idx_insert = idx_insert_last; 7758 } else { 7759 idx_insert = size; 7760 } 7761 } else { 7762 if (idx_insert_first != -1) { 7763 idx_insert = idx_insert_first + 1; 7764 } 7765 } 7766 7767 // always read samples from chain input buffer 7768 effect->setInBuffer(mInBuffer); 7769 7770 // if last effect in the chain, output samples to chain 7771 // output buffer, otherwise to chain input buffer 7772 if (idx_insert == size) { 7773 if (idx_insert != 0) { 7774 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7775 mEffects[idx_insert-1]->configure(); 7776 } 7777 effect->setOutBuffer(mOutBuffer); 7778 } else { 7779 effect->setOutBuffer(mInBuffer); 7780 } 7781 mEffects.insertAt(effect, idx_insert); 7782 7783 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7784 } 7785 effect->configure(); 7786 return NO_ERROR; 7787} 7788 7789// removeEffect_l() must be called with PlaybackThread::mLock held 7790size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7791{ 7792 Mutex::Autolock _l(mLock); 7793 size_t size = mEffects.size(); 7794 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7795 7796 for (size_t i = 0; i < size; i++) { 7797 if (effect == mEffects[i]) { 7798 // calling stop here will remove pre-processing effect from the audio HAL. 7799 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7800 // the middle of a read from audio HAL 7801 if (mEffects[i]->state() == EffectModule::ACTIVE || 7802 mEffects[i]->state() == EffectModule::STOPPING) { 7803 mEffects[i]->stop(); 7804 } 7805 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7806 delete[] effect->inBuffer(); 7807 } else { 7808 if (i == size - 1 && i != 0) { 7809 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7810 mEffects[i - 1]->configure(); 7811 } 7812 } 7813 mEffects.removeAt(i); 7814 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7815 break; 7816 } 7817 } 7818 7819 return mEffects.size(); 7820} 7821 7822// setDevice_l() must be called with PlaybackThread::mLock held 7823void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7824{ 7825 size_t size = mEffects.size(); 7826 for (size_t i = 0; i < size; i++) { 7827 mEffects[i]->setDevice(device); 7828 } 7829} 7830 7831// setMode_l() must be called with PlaybackThread::mLock held 7832void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7833{ 7834 size_t size = mEffects.size(); 7835 for (size_t i = 0; i < size; i++) { 7836 mEffects[i]->setMode(mode); 7837 } 7838} 7839 7840// setVolume_l() must be called with PlaybackThread::mLock held 7841bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7842{ 7843 uint32_t newLeft = *left; 7844 uint32_t newRight = *right; 7845 bool hasControl = false; 7846 int ctrlIdx = -1; 7847 size_t size = mEffects.size(); 7848 7849 // first update volume controller 7850 for (size_t i = size; i > 0; i--) { 7851 if (mEffects[i - 1]->isProcessEnabled() && 7852 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7853 ctrlIdx = i - 1; 7854 hasControl = true; 7855 break; 7856 } 7857 } 7858 7859 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7860 if (hasControl) { 7861 *left = mNewLeftVolume; 7862 *right = mNewRightVolume; 7863 } 7864 return hasControl; 7865 } 7866 7867 mVolumeCtrlIdx = ctrlIdx; 7868 mLeftVolume = newLeft; 7869 mRightVolume = newRight; 7870 7871 // second get volume update from volume controller 7872 if (ctrlIdx >= 0) { 7873 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7874 mNewLeftVolume = newLeft; 7875 mNewRightVolume = newRight; 7876 } 7877 // then indicate volume to all other effects in chain. 7878 // Pass altered volume to effects before volume controller 7879 // and requested volume to effects after controller 7880 uint32_t lVol = newLeft; 7881 uint32_t rVol = newRight; 7882 7883 for (size_t i = 0; i < size; i++) { 7884 if ((int)i == ctrlIdx) continue; 7885 // this also works for ctrlIdx == -1 when there is no volume controller 7886 if ((int)i > ctrlIdx) { 7887 lVol = *left; 7888 rVol = *right; 7889 } 7890 mEffects[i]->setVolume(&lVol, &rVol, false); 7891 } 7892 *left = newLeft; 7893 *right = newRight; 7894 7895 return hasControl; 7896} 7897 7898status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7899{ 7900 const size_t SIZE = 256; 7901 char buffer[SIZE]; 7902 String8 result; 7903 7904 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7905 result.append(buffer); 7906 7907 bool locked = tryLock(mLock); 7908 // failed to lock - AudioFlinger is probably deadlocked 7909 if (!locked) { 7910 result.append("\tCould not lock mutex:\n"); 7911 } 7912 7913 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7914 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7915 mEffects.size(), 7916 (uint32_t)mInBuffer, 7917 (uint32_t)mOutBuffer, 7918 mActiveTrackCnt); 7919 result.append(buffer); 7920 write(fd, result.string(), result.size()); 7921 7922 for (size_t i = 0; i < mEffects.size(); ++i) { 7923 sp<EffectModule> effect = mEffects[i]; 7924 if (effect != 0) { 7925 effect->dump(fd, args); 7926 } 7927 } 7928 7929 if (locked) { 7930 mLock.unlock(); 7931 } 7932 7933 return NO_ERROR; 7934} 7935 7936// must be called with ThreadBase::mLock held 7937void AudioFlinger::EffectChain::setEffectSuspended_l( 7938 const effect_uuid_t *type, bool suspend) 7939{ 7940 sp<SuspendedEffectDesc> desc; 7941 // use effect type UUID timelow as key as there is no real risk of identical 7942 // timeLow fields among effect type UUIDs. 7943 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7944 if (suspend) { 7945 if (index >= 0) { 7946 desc = mSuspendedEffects.valueAt(index); 7947 } else { 7948 desc = new SuspendedEffectDesc(); 7949 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7950 mSuspendedEffects.add(type->timeLow, desc); 7951 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7952 } 7953 if (desc->mRefCount++ == 0) { 7954 sp<EffectModule> effect = getEffectIfEnabled(type); 7955 if (effect != 0) { 7956 desc->mEffect = effect; 7957 effect->setSuspended(true); 7958 effect->setEnabled(false); 7959 } 7960 } 7961 } else { 7962 if (index < 0) { 7963 return; 7964 } 7965 desc = mSuspendedEffects.valueAt(index); 7966 if (desc->mRefCount <= 0) { 7967 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7968 desc->mRefCount = 1; 7969 } 7970 if (--desc->mRefCount == 0) { 7971 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7972 if (desc->mEffect != 0) { 7973 sp<EffectModule> effect = desc->mEffect.promote(); 7974 if (effect != 0) { 7975 effect->setSuspended(false); 7976 sp<EffectHandle> handle = effect->controlHandle(); 7977 if (handle != 0) { 7978 effect->setEnabled(handle->enabled()); 7979 } 7980 } 7981 desc->mEffect.clear(); 7982 } 7983 mSuspendedEffects.removeItemsAt(index); 7984 } 7985 } 7986} 7987 7988// must be called with ThreadBase::mLock held 7989void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7990{ 7991 sp<SuspendedEffectDesc> desc; 7992 7993 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7994 if (suspend) { 7995 if (index >= 0) { 7996 desc = mSuspendedEffects.valueAt(index); 7997 } else { 7998 desc = new SuspendedEffectDesc(); 7999 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8000 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8001 } 8002 if (desc->mRefCount++ == 0) { 8003 Vector< sp<EffectModule> > effects; 8004 getSuspendEligibleEffects(effects); 8005 for (size_t i = 0; i < effects.size(); i++) { 8006 setEffectSuspended_l(&effects[i]->desc().type, true); 8007 } 8008 } 8009 } else { 8010 if (index < 0) { 8011 return; 8012 } 8013 desc = mSuspendedEffects.valueAt(index); 8014 if (desc->mRefCount <= 0) { 8015 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8016 desc->mRefCount = 1; 8017 } 8018 if (--desc->mRefCount == 0) { 8019 Vector<const effect_uuid_t *> types; 8020 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8021 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8022 continue; 8023 } 8024 types.add(&mSuspendedEffects.valueAt(i)->mType); 8025 } 8026 for (size_t i = 0; i < types.size(); i++) { 8027 setEffectSuspended_l(types[i], false); 8028 } 8029 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8030 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8031 } 8032 } 8033} 8034 8035 8036// The volume effect is used for automated tests only 8037#ifndef OPENSL_ES_H_ 8038static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8039 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8040const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8041#endif //OPENSL_ES_H_ 8042 8043bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8044{ 8045 // auxiliary effects and visualizer are never suspended on output mix 8046 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8047 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8048 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8049 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8050 return false; 8051 } 8052 return true; 8053} 8054 8055void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8056{ 8057 effects.clear(); 8058 for (size_t i = 0; i < mEffects.size(); i++) { 8059 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8060 effects.add(mEffects[i]); 8061 } 8062 } 8063} 8064 8065sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8066 const effect_uuid_t *type) 8067{ 8068 sp<EffectModule> effect = getEffectFromType_l(type); 8069 return effect != 0 && effect->isEnabled() ? effect : 0; 8070} 8071 8072void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8073 bool enabled) 8074{ 8075 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8076 if (enabled) { 8077 if (index < 0) { 8078 // if the effect is not suspend check if all effects are suspended 8079 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8080 if (index < 0) { 8081 return; 8082 } 8083 if (!isEffectEligibleForSuspend(effect->desc())) { 8084 return; 8085 } 8086 setEffectSuspended_l(&effect->desc().type, enabled); 8087 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8088 if (index < 0) { 8089 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8090 return; 8091 } 8092 } 8093 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8094 effect->desc().type.timeLow); 8095 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8096 // if effect is requested to suspended but was not yet enabled, supend it now. 8097 if (desc->mEffect == 0) { 8098 desc->mEffect = effect; 8099 effect->setEnabled(false); 8100 effect->setSuspended(true); 8101 } 8102 } else { 8103 if (index < 0) { 8104 return; 8105 } 8106 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8107 effect->desc().type.timeLow); 8108 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8109 desc->mEffect.clear(); 8110 effect->setSuspended(false); 8111 } 8112} 8113 8114#undef LOG_TAG 8115#define LOG_TAG "AudioFlinger" 8116 8117// ---------------------------------------------------------------------------- 8118 8119status_t AudioFlinger::onTransact( 8120 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8121{ 8122 return BnAudioFlinger::onTransact(code, data, reply, flags); 8123} 8124 8125}; // namespace android 8126