AudioFlinger.cpp revision 5ce96d97feafc6989f6141bb2633eae3d87ddf28
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 1976bool AudioFlinger::MixerThread::threadLoop() 1977{ 1978 Vector< sp<Track> > tracksToRemove; 1979 mixer_state mixerStatus = MIXER_IDLE; 1980 nsecs_t standbyTime = systemTime(); 1981 size_t mixBufferSize = mFrameCount * mFrameSize; 1982 // FIXME: Relaxed timing because of a certain device that can't meet latency 1983 // Should be reduced to 2x after the vendor fixes the driver issue 1984 // increase threshold again due to low power audio mode. The way this warning threshold is 1985 // calculated and its usefulness should be reconsidered anyway. 1986 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1987 nsecs_t lastWarning = 0; 1988 bool longStandbyExit = false; 1989 uint32_t activeSleepTime = activeSleepTimeUs(); 1990 uint32_t idleSleepTime = idleSleepTimeUs(); 1991 uint32_t sleepTime = idleSleepTime; 1992 uint32_t sleepTimeShift = 0; 1993 Vector< sp<EffectChain> > effectChains; 1994 CpuStats cpuStats; 1995 1996 acquireWakeLock(); 1997 1998 while (!exitPending()) 1999 { 2000 cpuStats.sample(); 2001 processConfigEvents(); 2002 2003 mixerStatus = MIXER_IDLE; 2004 { // scope for mLock 2005 2006 Mutex::Autolock _l(mLock); 2007 2008 if (checkForNewParameters_l()) { 2009 mixBufferSize = mFrameCount * mFrameSize; 2010 // FIXME: Relaxed timing because of a certain device that can't meet latency 2011 // Should be reduced to 2x after the vendor fixes the driver issue 2012 // increase threshold again due to low power audio mode. The way this warning 2013 // threshold is calculated and its usefulness should be reconsidered anyway. 2014 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2015 activeSleepTime = activeSleepTimeUs(); 2016 idleSleepTime = idleSleepTimeUs(); 2017 } 2018 2019 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2020 2021 // put audio hardware into standby after short delay 2022 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2023 mSuspended)) { 2024 if (!mStandby) { 2025 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2026 mOutput->stream->common.standby(&mOutput->stream->common); 2027 mStandby = true; 2028 mBytesWritten = 0; 2029 } 2030 2031 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2032 // we're about to wait, flush the binder command buffer 2033 IPCThreadState::self()->flushCommands(); 2034 2035 if (exitPending()) break; 2036 2037 releaseWakeLock_l(); 2038 // wait until we have something to do... 2039 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 2040 mWaitWorkCV.wait(mLock); 2041 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 2042 acquireWakeLock_l(); 2043 2044 mPrevMixerStatus = MIXER_IDLE; 2045 if (!mMasterMute) { 2046 char value[PROPERTY_VALUE_MAX]; 2047 property_get("ro.audio.silent", value, "0"); 2048 if (atoi(value)) { 2049 ALOGD("Silence is golden"); 2050 setMasterMute_l(true); 2051 } 2052 } 2053 2054 standbyTime = systemTime() + mStandbyTimeInNsecs; 2055 sleepTime = idleSleepTime; 2056 sleepTimeShift = 0; 2057 continue; 2058 } 2059 } 2060 2061 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2062 2063 // prevent any changes in effect chain list and in each effect chain 2064 // during mixing and effect process as the audio buffers could be deleted 2065 // or modified if an effect is created or deleted 2066 lockEffectChains_l(effectChains); 2067 } 2068 2069 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2070 // obtain the presentation timestamp of the next output buffer 2071 int64_t pts; 2072 status_t status = INVALID_OPERATION; 2073 2074 if (NULL != mOutput->stream->get_next_write_timestamp) { 2075 status = mOutput->stream->get_next_write_timestamp( 2076 mOutput->stream, &pts); 2077 } 2078 2079 if (status != NO_ERROR) { 2080 pts = AudioBufferProvider::kInvalidPTS; 2081 } 2082 2083 // mix buffers... 2084 mAudioMixer->process(pts); 2085 // increase sleep time progressively when application underrun condition clears. 2086 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2087 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2088 // such that we would underrun the audio HAL. 2089 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2090 sleepTimeShift--; 2091 } 2092 sleepTime = 0; 2093 standbyTime = systemTime() + mStandbyTimeInNsecs; 2094 //TODO: delay standby when effects have a tail 2095 } else { 2096 // If no tracks are ready, sleep once for the duration of an output 2097 // buffer size, then write 0s to the output 2098 if (sleepTime == 0) { 2099 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2100 sleepTime = activeSleepTime >> sleepTimeShift; 2101 if (sleepTime < kMinThreadSleepTimeUs) { 2102 sleepTime = kMinThreadSleepTimeUs; 2103 } 2104 // reduce sleep time in case of consecutive application underruns to avoid 2105 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2106 // duration we would end up writing less data than needed by the audio HAL if 2107 // the condition persists. 2108 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2109 sleepTimeShift++; 2110 } 2111 } else { 2112 sleepTime = idleSleepTime; 2113 } 2114 } else if (mBytesWritten != 0 || 2115 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2116 memset (mMixBuffer, 0, mixBufferSize); 2117 sleepTime = 0; 2118 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2119 } 2120 // TODO add standby time extension fct of effect tail 2121 } 2122 2123 if (mSuspended) { 2124 sleepTime = suspendSleepTimeUs(); 2125 } 2126 // sleepTime == 0 means we must write to audio hardware 2127 if (sleepTime == 0) { 2128 for (size_t i = 0; i < effectChains.size(); i ++) { 2129 effectChains[i]->process_l(); 2130 } 2131 // enable changes in effect chain 2132 unlockEffectChains(effectChains); 2133 mLastWriteTime = systemTime(); 2134 mInWrite = true; 2135 mBytesWritten += mixBufferSize; 2136 2137 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2138 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2139 mNumWrites++; 2140 mInWrite = false; 2141 nsecs_t now = systemTime(); 2142 nsecs_t delta = now - mLastWriteTime; 2143 if (!mStandby && delta > maxPeriod) { 2144 mNumDelayedWrites++; 2145 if ((now - lastWarning) > kWarningThrottleNs) { 2146 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2147 ns2ms(delta), mNumDelayedWrites, this); 2148 lastWarning = now; 2149 } 2150 if (mStandby) { 2151 longStandbyExit = true; 2152 } 2153 } 2154 mStandby = false; 2155 } else { 2156 // enable changes in effect chain 2157 unlockEffectChains(effectChains); 2158 usleep(sleepTime); 2159 } 2160 2161 // finally let go of all our tracks, without the lock held 2162 // since we can't guarantee the destructors won't acquire that 2163 // same lock. 2164 tracksToRemove.clear(); 2165 2166 // Effect chains will be actually deleted here if they were removed from 2167 // mEffectChains list during mixing or effects processing 2168 effectChains.clear(); 2169 } 2170 2171 if (!mStandby) { 2172 mOutput->stream->common.standby(&mOutput->stream->common); 2173 } 2174 2175 releaseWakeLock(); 2176 2177 ALOGV("MixerThread %p exiting", this); 2178 return false; 2179} 2180 2181// prepareTracks_l() must be called with ThreadBase::mLock held 2182AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2183 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2184{ 2185 2186 mixer_state mixerStatus = MIXER_IDLE; 2187 // find out which tracks need to be processed 2188 size_t count = activeTracks.size(); 2189 size_t mixedTracks = 0; 2190 size_t tracksWithEffect = 0; 2191 2192 float masterVolume = mMasterVolume; 2193 bool masterMute = mMasterMute; 2194 2195 if (masterMute) { 2196 masterVolume = 0; 2197 } 2198 // Delegate master volume control to effect in output mix effect chain if needed 2199 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2200 if (chain != 0) { 2201 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2202 chain->setVolume_l(&v, &v); 2203 masterVolume = (float)((v + (1 << 23)) >> 24); 2204 chain.clear(); 2205 } 2206 2207 for (size_t i=0 ; i<count ; i++) { 2208 sp<Track> t = activeTracks[i].promote(); 2209 if (t == 0) continue; 2210 2211 // this const just means the local variable doesn't change 2212 Track* const track = t.get(); 2213 audio_track_cblk_t* cblk = track->cblk(); 2214 2215 // The first time a track is added we wait 2216 // for all its buffers to be filled before processing it 2217 int name = track->name(); 2218 // make sure that we have enough frames to mix one full buffer. 2219 // enforce this condition only once to enable draining the buffer in case the client 2220 // app does not call stop() and relies on underrun to stop: 2221 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2222 // during last round 2223 uint32_t minFrames = 1; 2224 if (!track->isStopped() && !track->isPausing() && 2225 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2226 if (t->sampleRate() == (int)mSampleRate) { 2227 minFrames = mFrameCount; 2228 } else { 2229 // +1 for rounding and +1 for additional sample needed for interpolation 2230 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2231 // add frames already consumed but not yet released by the resampler 2232 // because cblk->framesReady() will include these frames 2233 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2234 // the minimum track buffer size is normally twice the number of frames necessary 2235 // to fill one buffer and the resampler should not leave more than one buffer worth 2236 // of unreleased frames after each pass, but just in case... 2237 ALOG_ASSERT(minFrames <= cblk->frameCount); 2238 } 2239 } 2240 if ((track->framesReady() >= minFrames) && track->isReady() && 2241 !track->isPaused() && !track->isTerminated()) 2242 { 2243 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2244 2245 mixedTracks++; 2246 2247 // track->mainBuffer() != mMixBuffer means there is an effect chain 2248 // connected to the track 2249 chain.clear(); 2250 if (track->mainBuffer() != mMixBuffer) { 2251 chain = getEffectChain_l(track->sessionId()); 2252 // Delegate volume control to effect in track effect chain if needed 2253 if (chain != 0) { 2254 tracksWithEffect++; 2255 } else { 2256 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2257 name, track->sessionId()); 2258 } 2259 } 2260 2261 2262 int param = AudioMixer::VOLUME; 2263 if (track->mFillingUpStatus == Track::FS_FILLED) { 2264 // no ramp for the first volume setting 2265 track->mFillingUpStatus = Track::FS_ACTIVE; 2266 if (track->mState == TrackBase::RESUMING) { 2267 track->mState = TrackBase::ACTIVE; 2268 param = AudioMixer::RAMP_VOLUME; 2269 } 2270 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2271 } else if (cblk->server != 0) { 2272 // If the track is stopped before the first frame was mixed, 2273 // do not apply ramp 2274 param = AudioMixer::RAMP_VOLUME; 2275 } 2276 2277 // compute volume for this track 2278 uint32_t vl, vr, va; 2279 if (track->isMuted() || track->isPausing() || 2280 mStreamTypes[track->streamType()].mute) { 2281 vl = vr = va = 0; 2282 if (track->isPausing()) { 2283 track->setPaused(); 2284 } 2285 } else { 2286 2287 // read original volumes with volume control 2288 float typeVolume = mStreamTypes[track->streamType()].volume; 2289 float v = masterVolume * typeVolume; 2290 uint32_t vlr = cblk->getVolumeLR(); 2291 vl = vlr & 0xFFFF; 2292 vr = vlr >> 16; 2293 // track volumes come from shared memory, so can't be trusted and must be clamped 2294 if (vl > MAX_GAIN_INT) { 2295 ALOGV("Track left volume out of range: %04X", vl); 2296 vl = MAX_GAIN_INT; 2297 } 2298 if (vr > MAX_GAIN_INT) { 2299 ALOGV("Track right volume out of range: %04X", vr); 2300 vr = MAX_GAIN_INT; 2301 } 2302 // now apply the master volume and stream type volume 2303 vl = (uint32_t)(v * vl) << 12; 2304 vr = (uint32_t)(v * vr) << 12; 2305 // assuming master volume and stream type volume each go up to 1.0, 2306 // vl and vr are now in 8.24 format 2307 2308 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2309 // send level comes from shared memory and so may be corrupt 2310 if (sendLevel > MAX_GAIN_INT) { 2311 ALOGV("Track send level out of range: %04X", sendLevel); 2312 sendLevel = MAX_GAIN_INT; 2313 } 2314 va = (uint32_t)(v * sendLevel); 2315 } 2316 // Delegate volume control to effect in track effect chain if needed 2317 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2318 // Do not ramp volume if volume is controlled by effect 2319 param = AudioMixer::VOLUME; 2320 track->mHasVolumeController = true; 2321 } else { 2322 // force no volume ramp when volume controller was just disabled or removed 2323 // from effect chain to avoid volume spike 2324 if (track->mHasVolumeController) { 2325 param = AudioMixer::VOLUME; 2326 } 2327 track->mHasVolumeController = false; 2328 } 2329 2330 // Convert volumes from 8.24 to 4.12 format 2331 // This additional clamping is needed in case chain->setVolume_l() overshot 2332 vl = (vl + (1 << 11)) >> 12; 2333 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2334 vr = (vr + (1 << 11)) >> 12; 2335 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2336 2337 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2338 2339 // XXX: these things DON'T need to be done each time 2340 mAudioMixer->setBufferProvider(name, track); 2341 mAudioMixer->enable(name); 2342 2343 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2344 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2345 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2346 mAudioMixer->setParameter( 2347 name, 2348 AudioMixer::TRACK, 2349 AudioMixer::FORMAT, (void *)track->format()); 2350 mAudioMixer->setParameter( 2351 name, 2352 AudioMixer::TRACK, 2353 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2354 mAudioMixer->setParameter( 2355 name, 2356 AudioMixer::RESAMPLE, 2357 AudioMixer::SAMPLE_RATE, 2358 (void *)(cblk->sampleRate)); 2359 mAudioMixer->setParameter( 2360 name, 2361 AudioMixer::TRACK, 2362 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2363 mAudioMixer->setParameter( 2364 name, 2365 AudioMixer::TRACK, 2366 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2367 2368 // reset retry count 2369 track->mRetryCount = kMaxTrackRetries; 2370 // If one track is ready, set the mixer ready if: 2371 // - the mixer was not ready during previous round OR 2372 // - no other track is not ready 2373 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2374 mixerStatus != MIXER_TRACKS_ENABLED) { 2375 mixerStatus = MIXER_TRACKS_READY; 2376 } 2377 } else { 2378 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2379 if (track->isStopped()) { 2380 track->reset(); 2381 } 2382 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2383 // We have consumed all the buffers of this track. 2384 // Remove it from the list of active tracks. 2385 tracksToRemove->add(track); 2386 } else { 2387 // No buffers for this track. Give it a few chances to 2388 // fill a buffer, then remove it from active list. 2389 if (--(track->mRetryCount) <= 0) { 2390 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2391 tracksToRemove->add(track); 2392 // indicate to client process that the track was disabled because of underrun 2393 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2394 // If one track is not ready, mark the mixer also not ready if: 2395 // - the mixer was ready during previous round OR 2396 // - no other track is ready 2397 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2398 mixerStatus != MIXER_TRACKS_READY) { 2399 mixerStatus = MIXER_TRACKS_ENABLED; 2400 } 2401 } 2402 mAudioMixer->disable(name); 2403 } 2404 } 2405 2406 // remove all the tracks that need to be... 2407 count = tracksToRemove->size(); 2408 if (CC_UNLIKELY(count)) { 2409 for (size_t i=0 ; i<count ; i++) { 2410 const sp<Track>& track = tracksToRemove->itemAt(i); 2411 mActiveTracks.remove(track); 2412 if (track->mainBuffer() != mMixBuffer) { 2413 chain = getEffectChain_l(track->sessionId()); 2414 if (chain != 0) { 2415 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2416 chain->decActiveTrackCnt(); 2417 } 2418 } 2419 if (track->isTerminated()) { 2420 removeTrack_l(track); 2421 } 2422 } 2423 } 2424 2425 // mix buffer must be cleared if all tracks are connected to an 2426 // effect chain as in this case the mixer will not write to 2427 // mix buffer and track effects will accumulate into it 2428 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2429 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2430 } 2431 2432 mPrevMixerStatus = mixerStatus; 2433 return mixerStatus; 2434} 2435 2436void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2437{ 2438 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2439 this, streamType, mTracks.size()); 2440 Mutex::Autolock _l(mLock); 2441 2442 size_t size = mTracks.size(); 2443 for (size_t i = 0; i < size; i++) { 2444 sp<Track> t = mTracks[i]; 2445 if (t->streamType() == streamType) { 2446 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2447 t->mCblk->cv.signal(); 2448 } 2449 } 2450} 2451 2452void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2453{ 2454 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2455 this, streamType, valid); 2456 Mutex::Autolock _l(mLock); 2457 2458 mStreamTypes[streamType].valid = valid; 2459} 2460 2461// getTrackName_l() must be called with ThreadBase::mLock held 2462int AudioFlinger::MixerThread::getTrackName_l() 2463{ 2464 return mAudioMixer->getTrackName(); 2465} 2466 2467// deleteTrackName_l() must be called with ThreadBase::mLock held 2468void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2469{ 2470 ALOGV("remove track (%d) and delete from mixer", name); 2471 mAudioMixer->deleteTrackName(name); 2472} 2473 2474// checkForNewParameters_l() must be called with ThreadBase::mLock held 2475bool AudioFlinger::MixerThread::checkForNewParameters_l() 2476{ 2477 bool reconfig = false; 2478 2479 while (!mNewParameters.isEmpty()) { 2480 status_t status = NO_ERROR; 2481 String8 keyValuePair = mNewParameters[0]; 2482 AudioParameter param = AudioParameter(keyValuePair); 2483 int value; 2484 2485 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2486 reconfig = true; 2487 } 2488 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2489 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2490 status = BAD_VALUE; 2491 } else { 2492 reconfig = true; 2493 } 2494 } 2495 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2496 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2497 status = BAD_VALUE; 2498 } else { 2499 reconfig = true; 2500 } 2501 } 2502 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2503 // do not accept frame count changes if tracks are open as the track buffer 2504 // size depends on frame count and correct behavior would not be guaranteed 2505 // if frame count is changed after track creation 2506 if (!mTracks.isEmpty()) { 2507 status = INVALID_OPERATION; 2508 } else { 2509 reconfig = true; 2510 } 2511 } 2512 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2513 // when changing the audio output device, call addBatteryData to notify 2514 // the change 2515 if ((int)mDevice != value) { 2516 uint32_t params = 0; 2517 // check whether speaker is on 2518 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2519 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2520 } 2521 2522 int deviceWithoutSpeaker 2523 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2524 // check if any other device (except speaker) is on 2525 if (value & deviceWithoutSpeaker ) { 2526 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2527 } 2528 2529 if (params != 0) { 2530 addBatteryData(params); 2531 } 2532 } 2533 2534 // forward device change to effects that have requested to be 2535 // aware of attached audio device. 2536 mDevice = (uint32_t)value; 2537 for (size_t i = 0; i < mEffectChains.size(); i++) { 2538 mEffectChains[i]->setDevice_l(mDevice); 2539 } 2540 } 2541 2542 if (status == NO_ERROR) { 2543 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2544 keyValuePair.string()); 2545 if (!mStandby && status == INVALID_OPERATION) { 2546 mOutput->stream->common.standby(&mOutput->stream->common); 2547 mStandby = true; 2548 mBytesWritten = 0; 2549 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2550 keyValuePair.string()); 2551 } 2552 if (status == NO_ERROR && reconfig) { 2553 delete mAudioMixer; 2554 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2555 mAudioMixer = NULL; 2556 readOutputParameters(); 2557 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2558 for (size_t i = 0; i < mTracks.size() ; i++) { 2559 int name = getTrackName_l(); 2560 if (name < 0) break; 2561 mTracks[i]->mName = name; 2562 // limit track sample rate to 2 x new output sample rate 2563 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2564 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2565 } 2566 } 2567 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2568 } 2569 } 2570 2571 mNewParameters.removeAt(0); 2572 2573 mParamStatus = status; 2574 mParamCond.signal(); 2575 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2576 // already timed out waiting for the status and will never signal the condition. 2577 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2578 } 2579 return reconfig; 2580} 2581 2582status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2583{ 2584 const size_t SIZE = 256; 2585 char buffer[SIZE]; 2586 String8 result; 2587 2588 PlaybackThread::dumpInternals(fd, args); 2589 2590 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2591 result.append(buffer); 2592 write(fd, result.string(), result.size()); 2593 return NO_ERROR; 2594} 2595 2596uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2597{ 2598 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2599} 2600 2601uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2602{ 2603 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2604} 2605 2606// ---------------------------------------------------------------------------- 2607AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2608 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2609 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2610 // mLeftVolFloat, mRightVolFloat 2611 // mLeftVolShort, mRightVolShort 2612{ 2613} 2614 2615AudioFlinger::DirectOutputThread::~DirectOutputThread() 2616{ 2617} 2618 2619void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2620{ 2621 // Do not apply volume on compressed audio 2622 if (!audio_is_linear_pcm(mFormat)) { 2623 return; 2624 } 2625 2626 // convert to signed 16 bit before volume calculation 2627 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2628 size_t count = mFrameCount * mChannelCount; 2629 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2630 int16_t *dst = mMixBuffer + count-1; 2631 while(count--) { 2632 *dst-- = (int16_t)(*src--^0x80) << 8; 2633 } 2634 } 2635 2636 size_t frameCount = mFrameCount; 2637 int16_t *out = mMixBuffer; 2638 if (ramp) { 2639 if (mChannelCount == 1) { 2640 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2641 int32_t vlInc = d / (int32_t)frameCount; 2642 int32_t vl = ((int32_t)mLeftVolShort << 16); 2643 do { 2644 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2645 out++; 2646 vl += vlInc; 2647 } while (--frameCount); 2648 2649 } else { 2650 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2651 int32_t vlInc = d / (int32_t)frameCount; 2652 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2653 int32_t vrInc = d / (int32_t)frameCount; 2654 int32_t vl = ((int32_t)mLeftVolShort << 16); 2655 int32_t vr = ((int32_t)mRightVolShort << 16); 2656 do { 2657 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2658 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2659 out += 2; 2660 vl += vlInc; 2661 vr += vrInc; 2662 } while (--frameCount); 2663 } 2664 } else { 2665 if (mChannelCount == 1) { 2666 do { 2667 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2668 out++; 2669 } while (--frameCount); 2670 } else { 2671 do { 2672 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2673 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2674 out += 2; 2675 } while (--frameCount); 2676 } 2677 } 2678 2679 // convert back to unsigned 8 bit after volume calculation 2680 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2681 size_t count = mFrameCount * mChannelCount; 2682 int16_t *src = mMixBuffer; 2683 uint8_t *dst = (uint8_t *)mMixBuffer; 2684 while(count--) { 2685 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2686 } 2687 } 2688 2689 mLeftVolShort = leftVol; 2690 mRightVolShort = rightVol; 2691} 2692 2693bool AudioFlinger::DirectOutputThread::threadLoop() 2694{ 2695 mixer_state mixerStatus = MIXER_IDLE; 2696 sp<Track> trackToRemove; 2697 sp<Track> activeTrack; 2698 nsecs_t standbyTime = systemTime(); 2699 size_t mixBufferSize = mFrameCount*mFrameSize; 2700 uint32_t activeSleepTime = activeSleepTimeUs(); 2701 uint32_t idleSleepTime = idleSleepTimeUs(); 2702 uint32_t sleepTime = idleSleepTime; 2703 // use shorter standby delay as on normal output to release 2704 // hardware resources as soon as possible 2705 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2706 2707 acquireWakeLock(); 2708 2709 while (!exitPending()) 2710 { 2711 bool rampVolume; 2712 uint16_t leftVol; 2713 uint16_t rightVol; 2714 Vector< sp<EffectChain> > effectChains; 2715 2716 processConfigEvents(); 2717 2718 mixerStatus = MIXER_IDLE; 2719 2720 { // scope for the mLock 2721 2722 Mutex::Autolock _l(mLock); 2723 2724 if (checkForNewParameters_l()) { 2725 mixBufferSize = mFrameCount*mFrameSize; 2726 activeSleepTime = activeSleepTimeUs(); 2727 idleSleepTime = idleSleepTimeUs(); 2728 standbyDelay = microseconds(activeSleepTime*2); 2729 } 2730 2731 // put audio hardware into standby after short delay 2732 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2733 mSuspended)) { 2734 // wait until we have something to do... 2735 if (!mStandby) { 2736 ALOGV("Audio hardware entering standby, mixer %p", this); 2737 mOutput->stream->common.standby(&mOutput->stream->common); 2738 mStandby = true; 2739 mBytesWritten = 0; 2740 } 2741 2742 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2743 // we're about to wait, flush the binder command buffer 2744 IPCThreadState::self()->flushCommands(); 2745 2746 if (exitPending()) break; 2747 2748 releaseWakeLock_l(); 2749 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2750 mWaitWorkCV.wait(mLock); 2751 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2752 acquireWakeLock_l(); 2753 2754 if (!mMasterMute) { 2755 char value[PROPERTY_VALUE_MAX]; 2756 property_get("ro.audio.silent", value, "0"); 2757 if (atoi(value)) { 2758 ALOGD("Silence is golden"); 2759 setMasterMute_l(true); 2760 } 2761 } 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("DirectOutputThread %p exiting", this); 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 mixer_state mixerStatus = MIXER_IDLE; 3093 nsecs_t standbyTime = systemTime(); 3094 size_t mixBufferSize = mFrameCount*mFrameSize; 3095 SortedVector< sp<OutputTrack> > outputTracks; 3096 uint32_t writeFrames = 0; 3097 uint32_t activeSleepTime = activeSleepTimeUs(); 3098 uint32_t idleSleepTime = idleSleepTimeUs(); 3099 uint32_t sleepTime = idleSleepTime; 3100 Vector< sp<EffectChain> > effectChains; 3101 3102 acquireWakeLock(); 3103 3104 while (!exitPending()) 3105 { 3106 processConfigEvents(); 3107 3108 mixerStatus = MIXER_IDLE; 3109 { // scope for the mLock 3110 3111 Mutex::Autolock _l(mLock); 3112 3113 if (checkForNewParameters_l()) { 3114 mixBufferSize = mFrameCount*mFrameSize; 3115 updateWaitTime(); 3116 activeSleepTime = activeSleepTimeUs(); 3117 idleSleepTime = idleSleepTimeUs(); 3118 } 3119 3120 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3121 3122 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3123 outputTracks.add(mOutputTracks[i]); 3124 } 3125 3126 // put audio hardware into standby after short delay 3127 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3128 mSuspended)) { 3129 if (!mStandby) { 3130 for (size_t i = 0; i < outputTracks.size(); i++) { 3131 outputTracks[i]->stop(); 3132 } 3133 mStandby = true; 3134 mBytesWritten = 0; 3135 } 3136 3137 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3138 // we're about to wait, flush the binder command buffer 3139 IPCThreadState::self()->flushCommands(); 3140 outputTracks.clear(); 3141 3142 if (exitPending()) break; 3143 3144 releaseWakeLock_l(); 3145 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3146 mWaitWorkCV.wait(mLock); 3147 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3148 acquireWakeLock_l(); 3149 3150 mPrevMixerStatus = MIXER_IDLE; 3151 if (!mMasterMute) { 3152 char value[PROPERTY_VALUE_MAX]; 3153 property_get("ro.audio.silent", value, "0"); 3154 if (atoi(value)) { 3155 ALOGD("Silence is golden"); 3156 setMasterMute_l(true); 3157 } 3158 } 3159 3160 standbyTime = systemTime() + mStandbyTimeInNsecs; 3161 sleepTime = idleSleepTime; 3162 continue; 3163 } 3164 } 3165 3166 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3167 3168 // prevent any changes in effect chain list and in each effect chain 3169 // during mixing and effect process as the audio buffers could be deleted 3170 // or modified if an effect is created or deleted 3171 lockEffectChains_l(effectChains); 3172 } 3173 3174 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3175 // mix buffers... 3176 if (outputsReady(outputTracks)) { 3177 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3178 } else { 3179 memset(mMixBuffer, 0, mixBufferSize); 3180 } 3181 sleepTime = 0; 3182 writeFrames = mFrameCount; 3183 } else { 3184 if (sleepTime == 0) { 3185 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3186 sleepTime = activeSleepTime; 3187 } else { 3188 sleepTime = idleSleepTime; 3189 } 3190 } else if (mBytesWritten != 0) { 3191 // flush remaining overflow buffers in output tracks 3192 for (size_t i = 0; i < outputTracks.size(); i++) { 3193 if (outputTracks[i]->isActive()) { 3194 sleepTime = 0; 3195 writeFrames = 0; 3196 memset(mMixBuffer, 0, mixBufferSize); 3197 break; 3198 } 3199 } 3200 } 3201 } 3202 3203 if (mSuspended) { 3204 sleepTime = suspendSleepTimeUs(); 3205 } 3206 // sleepTime == 0 means we must write to audio hardware 3207 if (sleepTime == 0) { 3208 for (size_t i = 0; i < effectChains.size(); i ++) { 3209 effectChains[i]->process_l(); 3210 } 3211 // enable changes in effect chain 3212 unlockEffectChains(effectChains); 3213 3214 standbyTime = systemTime() + mStandbyTimeInNsecs; 3215 for (size_t i = 0; i < outputTracks.size(); i++) { 3216 outputTracks[i]->write(mMixBuffer, writeFrames); 3217 } 3218 mStandby = false; 3219 mBytesWritten += mixBufferSize; 3220 } else { 3221 // enable changes in effect chain 3222 unlockEffectChains(effectChains); 3223 usleep(sleepTime); 3224 } 3225 3226 // finally let go of all our tracks, without the lock held 3227 // since we can't guarantee the destructors won't acquire that 3228 // same lock. 3229 tracksToRemove.clear(); 3230 outputTracks.clear(); 3231 3232 // Effect chains will be actually deleted here if they were removed from 3233 // mEffectChains list during mixing or effects processing 3234 effectChains.clear(); 3235 } 3236 3237 releaseWakeLock(); 3238 3239 return false; 3240} 3241 3242void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3243{ 3244 // FIXME explain this formula 3245 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3246 OutputTrack *outputTrack = new OutputTrack(thread, 3247 this, 3248 mSampleRate, 3249 mFormat, 3250 mChannelMask, 3251 frameCount); 3252 if (outputTrack->cblk() != NULL) { 3253 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3254 mOutputTracks.add(outputTrack); 3255 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3256 updateWaitTime(); 3257 } 3258} 3259 3260void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3261{ 3262 Mutex::Autolock _l(mLock); 3263 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3264 if (mOutputTracks[i]->thread() == thread) { 3265 mOutputTracks[i]->destroy(); 3266 mOutputTracks.removeAt(i); 3267 updateWaitTime(); 3268 return; 3269 } 3270 } 3271 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3272} 3273 3274void AudioFlinger::DuplicatingThread::updateWaitTime() 3275{ 3276 mWaitTimeMs = UINT_MAX; 3277 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3278 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3279 if (strong != 0) { 3280 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3281 if (waitTimeMs < mWaitTimeMs) { 3282 mWaitTimeMs = waitTimeMs; 3283 } 3284 } 3285 } 3286} 3287 3288 3289bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3290{ 3291 for (size_t i = 0; i < outputTracks.size(); i++) { 3292 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3293 if (thread == 0) { 3294 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3295 return false; 3296 } 3297 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3298 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3299 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3300 return false; 3301 } 3302 } 3303 return true; 3304} 3305 3306uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3307{ 3308 return (mWaitTimeMs * 1000) / 2; 3309} 3310 3311// ---------------------------------------------------------------------------- 3312 3313// TrackBase constructor must be called with AudioFlinger::mLock held 3314AudioFlinger::ThreadBase::TrackBase::TrackBase( 3315 ThreadBase *thread, 3316 const sp<Client>& client, 3317 uint32_t sampleRate, 3318 audio_format_t format, 3319 uint32_t channelMask, 3320 int frameCount, 3321 const sp<IMemory>& sharedBuffer, 3322 int sessionId) 3323 : RefBase(), 3324 mThread(thread), 3325 mClient(client), 3326 mCblk(NULL), 3327 // mBuffer 3328 // mBufferEnd 3329 mFrameCount(0), 3330 mState(IDLE), 3331 mFormat(format), 3332 mStepServerFailed(false), 3333 mSessionId(sessionId) 3334 // mChannelCount 3335 // mChannelMask 3336{ 3337 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3338 3339 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3340 size_t size = sizeof(audio_track_cblk_t); 3341 uint8_t channelCount = popcount(channelMask); 3342 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3343 if (sharedBuffer == 0) { 3344 size += bufferSize; 3345 } 3346 3347 if (client != NULL) { 3348 mCblkMemory = client->heap()->allocate(size); 3349 if (mCblkMemory != 0) { 3350 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3351 if (mCblk != NULL) { // construct the shared structure in-place. 3352 new(mCblk) audio_track_cblk_t(); 3353 // clear all buffers 3354 mCblk->frameCount = frameCount; 3355 mCblk->sampleRate = sampleRate; 3356 mChannelCount = channelCount; 3357 mChannelMask = channelMask; 3358 if (sharedBuffer == 0) { 3359 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3360 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3361 // Force underrun condition to avoid false underrun callback until first data is 3362 // written to buffer (other flags are cleared) 3363 mCblk->flags = CBLK_UNDERRUN_ON; 3364 } else { 3365 mBuffer = sharedBuffer->pointer(); 3366 } 3367 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3368 } 3369 } else { 3370 ALOGE("not enough memory for AudioTrack size=%u", size); 3371 client->heap()->dump("AudioTrack"); 3372 return; 3373 } 3374 } else { 3375 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3376 // construct the shared structure in-place. 3377 new(mCblk) audio_track_cblk_t(); 3378 // clear all buffers 3379 mCblk->frameCount = frameCount; 3380 mCblk->sampleRate = sampleRate; 3381 mChannelCount = channelCount; 3382 mChannelMask = channelMask; 3383 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3384 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3385 // Force underrun condition to avoid false underrun callback until first data is 3386 // written to buffer (other flags are cleared) 3387 mCblk->flags = CBLK_UNDERRUN_ON; 3388 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3389 } 3390} 3391 3392AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3393{ 3394 if (mCblk != NULL) { 3395 if (mClient == 0) { 3396 delete mCblk; 3397 } else { 3398 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3399 } 3400 } 3401 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3402 if (mClient != 0) { 3403 // Client destructor must run with AudioFlinger mutex locked 3404 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3405 // If the client's reference count drops to zero, the associated destructor 3406 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3407 // relying on the automatic clear() at end of scope. 3408 mClient.clear(); 3409 } 3410} 3411 3412void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3413{ 3414 buffer->raw = NULL; 3415 mFrameCount = buffer->frameCount; 3416 step(); 3417 buffer->frameCount = 0; 3418} 3419 3420bool AudioFlinger::ThreadBase::TrackBase::step() { 3421 bool result; 3422 audio_track_cblk_t* cblk = this->cblk(); 3423 3424 result = cblk->stepServer(mFrameCount); 3425 if (!result) { 3426 ALOGV("stepServer failed acquiring cblk mutex"); 3427 mStepServerFailed = true; 3428 } 3429 return result; 3430} 3431 3432void AudioFlinger::ThreadBase::TrackBase::reset() { 3433 audio_track_cblk_t* cblk = this->cblk(); 3434 3435 cblk->user = 0; 3436 cblk->server = 0; 3437 cblk->userBase = 0; 3438 cblk->serverBase = 0; 3439 mStepServerFailed = false; 3440 ALOGV("TrackBase::reset"); 3441} 3442 3443int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3444 return (int)mCblk->sampleRate; 3445} 3446 3447void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3448 audio_track_cblk_t* cblk = this->cblk(); 3449 size_t frameSize = cblk->frameSize; 3450 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3451 int8_t *bufferEnd = bufferStart + frames * frameSize; 3452 3453 // Check validity of returned pointer in case the track control block would have been corrupted. 3454 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3455 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3456 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3457 server %d, serverBase %d, user %d, userBase %d", 3458 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3459 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3460 return NULL; 3461 } 3462 3463 return bufferStart; 3464} 3465 3466// ---------------------------------------------------------------------------- 3467 3468// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3469AudioFlinger::PlaybackThread::Track::Track( 3470 PlaybackThread *thread, 3471 const sp<Client>& client, 3472 audio_stream_type_t streamType, 3473 uint32_t sampleRate, 3474 audio_format_t format, 3475 uint32_t channelMask, 3476 int frameCount, 3477 const sp<IMemory>& sharedBuffer, 3478 int sessionId) 3479 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3480 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3481 mAuxEffectId(0), mHasVolumeController(false) 3482{ 3483 if (mCblk != NULL) { 3484 if (thread != NULL) { 3485 mName = thread->getTrackName_l(); 3486 mMainBuffer = thread->mixBuffer(); 3487 } 3488 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3489 if (mName < 0) { 3490 ALOGE("no more track names available"); 3491 } 3492 mStreamType = streamType; 3493 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3494 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3495 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3496 } 3497} 3498 3499AudioFlinger::PlaybackThread::Track::~Track() 3500{ 3501 ALOGV("PlaybackThread::Track destructor"); 3502 sp<ThreadBase> thread = mThread.promote(); 3503 if (thread != 0) { 3504 Mutex::Autolock _l(thread->mLock); 3505 mState = TERMINATED; 3506 } 3507} 3508 3509void AudioFlinger::PlaybackThread::Track::destroy() 3510{ 3511 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3512 // by removing it from mTracks vector, so there is a risk that this Tracks's 3513 // destructor is called. As the destructor needs to lock mLock, 3514 // we must acquire a strong reference on this Track before locking mLock 3515 // here so that the destructor is called only when exiting this function. 3516 // On the other hand, as long as Track::destroy() is only called by 3517 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3518 // this Track with its member mTrack. 3519 sp<Track> keep(this); 3520 { // scope for mLock 3521 sp<ThreadBase> thread = mThread.promote(); 3522 if (thread != 0) { 3523 if (!isOutputTrack()) { 3524 if (mState == ACTIVE || mState == RESUMING) { 3525 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3526 3527 // to track the speaker usage 3528 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3529 } 3530 AudioSystem::releaseOutput(thread->id()); 3531 } 3532 Mutex::Autolock _l(thread->mLock); 3533 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3534 playbackThread->destroyTrack_l(this); 3535 } 3536 } 3537} 3538 3539void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3540{ 3541 uint32_t vlr = mCblk->getVolumeLR(); 3542 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", 3543 mName - AudioMixer::TRACK0, 3544 (mClient == 0) ? getpid_cached : mClient->pid(), 3545 mStreamType, 3546 mFormat, 3547 mChannelMask, 3548 mSessionId, 3549 mFrameCount, 3550 mState, 3551 mMute, 3552 mFillingUpStatus, 3553 mCblk->sampleRate, 3554 vlr & 0xFFFF, 3555 vlr >> 16, 3556 mCblk->server, 3557 mCblk->user, 3558 (int)mMainBuffer, 3559 (int)mAuxBuffer); 3560} 3561 3562status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3563 AudioBufferProvider::Buffer* buffer, int64_t pts) 3564{ 3565 audio_track_cblk_t* cblk = this->cblk(); 3566 uint32_t framesReady; 3567 uint32_t framesReq = buffer->frameCount; 3568 3569 // Check if last stepServer failed, try to step now 3570 if (mStepServerFailed) { 3571 if (!step()) goto getNextBuffer_exit; 3572 ALOGV("stepServer recovered"); 3573 mStepServerFailed = false; 3574 } 3575 3576 framesReady = cblk->framesReady(); 3577 3578 if (CC_LIKELY(framesReady)) { 3579 uint32_t s = cblk->server; 3580 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3581 3582 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3583 if (framesReq > framesReady) { 3584 framesReq = framesReady; 3585 } 3586 if (s + framesReq > bufferEnd) { 3587 framesReq = bufferEnd - s; 3588 } 3589 3590 buffer->raw = getBuffer(s, framesReq); 3591 if (buffer->raw == NULL) goto getNextBuffer_exit; 3592 3593 buffer->frameCount = framesReq; 3594 return NO_ERROR; 3595 } 3596 3597getNextBuffer_exit: 3598 buffer->raw = NULL; 3599 buffer->frameCount = 0; 3600 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3601 return NOT_ENOUGH_DATA; 3602} 3603 3604uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3605 return mCblk->framesReady(); 3606} 3607 3608bool AudioFlinger::PlaybackThread::Track::isReady() const { 3609 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3610 3611 if (framesReady() >= mCblk->frameCount || 3612 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3613 mFillingUpStatus = FS_FILLED; 3614 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3615 return true; 3616 } 3617 return false; 3618} 3619 3620status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3621{ 3622 status_t status = NO_ERROR; 3623 ALOGV("start(%d), calling pid %d session %d tid %d", 3624 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3625 sp<ThreadBase> thread = mThread.promote(); 3626 if (thread != 0) { 3627 Mutex::Autolock _l(thread->mLock); 3628 track_state state = mState; 3629 // here the track could be either new, or restarted 3630 // in both cases "unstop" the track 3631 if (mState == PAUSED) { 3632 mState = TrackBase::RESUMING; 3633 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3634 } else { 3635 mState = TrackBase::ACTIVE; 3636 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3637 } 3638 3639 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3640 thread->mLock.unlock(); 3641 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3642 thread->mLock.lock(); 3643 3644 // to track the speaker usage 3645 if (status == NO_ERROR) { 3646 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3647 } 3648 } 3649 if (status == NO_ERROR) { 3650 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3651 playbackThread->addTrack_l(this); 3652 } else { 3653 mState = state; 3654 } 3655 } else { 3656 status = BAD_VALUE; 3657 } 3658 return status; 3659} 3660 3661void AudioFlinger::PlaybackThread::Track::stop() 3662{ 3663 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3664 sp<ThreadBase> thread = mThread.promote(); 3665 if (thread != 0) { 3666 Mutex::Autolock _l(thread->mLock); 3667 track_state state = mState; 3668 if (mState > STOPPED) { 3669 mState = STOPPED; 3670 // If the track is not active (PAUSED and buffers full), flush buffers 3671 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3672 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3673 reset(); 3674 } 3675 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3676 } 3677 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3678 thread->mLock.unlock(); 3679 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3680 thread->mLock.lock(); 3681 3682 // to track the speaker usage 3683 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3684 } 3685 } 3686} 3687 3688void AudioFlinger::PlaybackThread::Track::pause() 3689{ 3690 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3691 sp<ThreadBase> thread = mThread.promote(); 3692 if (thread != 0) { 3693 Mutex::Autolock _l(thread->mLock); 3694 if (mState == ACTIVE || mState == RESUMING) { 3695 mState = PAUSING; 3696 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3697 if (!isOutputTrack()) { 3698 thread->mLock.unlock(); 3699 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3700 thread->mLock.lock(); 3701 3702 // to track the speaker usage 3703 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3704 } 3705 } 3706 } 3707} 3708 3709void AudioFlinger::PlaybackThread::Track::flush() 3710{ 3711 ALOGV("flush(%d)", mName); 3712 sp<ThreadBase> thread = mThread.promote(); 3713 if (thread != 0) { 3714 Mutex::Autolock _l(thread->mLock); 3715 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3716 return; 3717 } 3718 // No point remaining in PAUSED state after a flush => go to 3719 // STOPPED state 3720 mState = STOPPED; 3721 3722 // do not reset the track if it is still in the process of being stopped or paused. 3723 // this will be done by prepareTracks_l() when the track is stopped. 3724 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3725 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3726 reset(); 3727 } 3728 } 3729} 3730 3731void AudioFlinger::PlaybackThread::Track::reset() 3732{ 3733 // Do not reset twice to avoid discarding data written just after a flush and before 3734 // the audioflinger thread detects the track is stopped. 3735 if (!mResetDone) { 3736 TrackBase::reset(); 3737 // Force underrun condition to avoid false underrun callback until first data is 3738 // written to buffer 3739 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3740 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3741 mFillingUpStatus = FS_FILLING; 3742 mResetDone = true; 3743 } 3744} 3745 3746void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3747{ 3748 mMute = muted; 3749} 3750 3751status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3752{ 3753 status_t status = DEAD_OBJECT; 3754 sp<ThreadBase> thread = mThread.promote(); 3755 if (thread != 0) { 3756 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3757 status = playbackThread->attachAuxEffect(this, EffectId); 3758 } 3759 return status; 3760} 3761 3762void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3763{ 3764 mAuxEffectId = EffectId; 3765 mAuxBuffer = buffer; 3766} 3767 3768// timed audio tracks 3769 3770sp<AudioFlinger::PlaybackThread::TimedTrack> 3771AudioFlinger::PlaybackThread::TimedTrack::create( 3772 PlaybackThread *thread, 3773 const sp<Client>& client, 3774 audio_stream_type_t streamType, 3775 uint32_t sampleRate, 3776 audio_format_t format, 3777 uint32_t channelMask, 3778 int frameCount, 3779 const sp<IMemory>& sharedBuffer, 3780 int sessionId) { 3781 if (!client->reserveTimedTrack()) 3782 return NULL; 3783 3784 sp<TimedTrack> track = new TimedTrack( 3785 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3786 sharedBuffer, sessionId); 3787 3788 if (track == NULL) { 3789 client->releaseTimedTrack(); 3790 return NULL; 3791 } 3792 3793 return track; 3794} 3795 3796AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3797 PlaybackThread *thread, 3798 const sp<Client>& client, 3799 audio_stream_type_t streamType, 3800 uint32_t sampleRate, 3801 audio_format_t format, 3802 uint32_t channelMask, 3803 int frameCount, 3804 const sp<IMemory>& sharedBuffer, 3805 int sessionId) 3806 : Track(thread, client, streamType, sampleRate, format, channelMask, 3807 frameCount, sharedBuffer, sessionId), 3808 mTimedSilenceBuffer(NULL), 3809 mTimedSilenceBufferSize(0), 3810 mTimedAudioOutputOnTime(false), 3811 mMediaTimeTransformValid(false) 3812{ 3813 LocalClock lc; 3814 mLocalTimeFreq = lc.getLocalFreq(); 3815 3816 mLocalTimeToSampleTransform.a_zero = 0; 3817 mLocalTimeToSampleTransform.b_zero = 0; 3818 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3819 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3820 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3821 &mLocalTimeToSampleTransform.a_to_b_denom); 3822} 3823 3824AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3825 mClient->releaseTimedTrack(); 3826 delete [] mTimedSilenceBuffer; 3827} 3828 3829status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3830 size_t size, sp<IMemory>* buffer) { 3831 3832 Mutex::Autolock _l(mTimedBufferQueueLock); 3833 3834 trimTimedBufferQueue_l(); 3835 3836 // lazily initialize the shared memory heap for timed buffers 3837 if (mTimedMemoryDealer == NULL) { 3838 const int kTimedBufferHeapSize = 512 << 10; 3839 3840 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3841 "AudioFlingerTimed"); 3842 if (mTimedMemoryDealer == NULL) 3843 return NO_MEMORY; 3844 } 3845 3846 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3847 if (newBuffer == NULL) { 3848 newBuffer = mTimedMemoryDealer->allocate(size); 3849 if (newBuffer == NULL) 3850 return NO_MEMORY; 3851 } 3852 3853 *buffer = newBuffer; 3854 return NO_ERROR; 3855} 3856 3857// caller must hold mTimedBufferQueueLock 3858void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3859 int64_t mediaTimeNow; 3860 { 3861 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3862 if (!mMediaTimeTransformValid) 3863 return; 3864 3865 int64_t targetTimeNow; 3866 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3867 ? mCCHelper.getCommonTime(&targetTimeNow) 3868 : mCCHelper.getLocalTime(&targetTimeNow); 3869 3870 if (OK != res) 3871 return; 3872 3873 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3874 &mediaTimeNow)) { 3875 return; 3876 } 3877 } 3878 3879 size_t trimIndex; 3880 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3881 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3882 break; 3883 } 3884 3885 if (trimIndex) { 3886 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3887 } 3888} 3889 3890status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3891 const sp<IMemory>& buffer, int64_t pts) { 3892 3893 { 3894 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3895 if (!mMediaTimeTransformValid) 3896 return INVALID_OPERATION; 3897 } 3898 3899 Mutex::Autolock _l(mTimedBufferQueueLock); 3900 3901 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3902 3903 return NO_ERROR; 3904} 3905 3906status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3907 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3908 3909 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3910 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3911 target); 3912 3913 if (!(target == TimedAudioTrack::LOCAL_TIME || 3914 target == TimedAudioTrack::COMMON_TIME)) { 3915 return BAD_VALUE; 3916 } 3917 3918 Mutex::Autolock lock(mMediaTimeTransformLock); 3919 mMediaTimeTransform = xform; 3920 mMediaTimeTransformTarget = target; 3921 mMediaTimeTransformValid = true; 3922 3923 return NO_ERROR; 3924} 3925 3926#define min(a, b) ((a) < (b) ? (a) : (b)) 3927 3928// implementation of getNextBuffer for tracks whose buffers have timestamps 3929status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3930 AudioBufferProvider::Buffer* buffer, int64_t pts) 3931{ 3932 if (pts == AudioBufferProvider::kInvalidPTS) { 3933 buffer->raw = 0; 3934 buffer->frameCount = 0; 3935 return INVALID_OPERATION; 3936 } 3937 3938 Mutex::Autolock _l(mTimedBufferQueueLock); 3939 3940 while (true) { 3941 3942 // if we have no timed buffers, then fail 3943 if (mTimedBufferQueue.isEmpty()) { 3944 buffer->raw = 0; 3945 buffer->frameCount = 0; 3946 return NOT_ENOUGH_DATA; 3947 } 3948 3949 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3950 3951 // calculate the PTS of the head of the timed buffer queue expressed in 3952 // local time 3953 int64_t headLocalPTS; 3954 { 3955 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3956 3957 assert(mMediaTimeTransformValid); 3958 3959 if (mMediaTimeTransform.a_to_b_denom == 0) { 3960 // the transform represents a pause, so yield silence 3961 timedYieldSilence(buffer->frameCount, buffer); 3962 return NO_ERROR; 3963 } 3964 3965 int64_t transformedPTS; 3966 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3967 &transformedPTS)) { 3968 // the transform failed. this shouldn't happen, but if it does 3969 // then just drop this buffer 3970 ALOGW("timedGetNextBuffer transform failed"); 3971 buffer->raw = 0; 3972 buffer->frameCount = 0; 3973 mTimedBufferQueue.removeAt(0); 3974 return NO_ERROR; 3975 } 3976 3977 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3978 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3979 &headLocalPTS)) { 3980 buffer->raw = 0; 3981 buffer->frameCount = 0; 3982 return INVALID_OPERATION; 3983 } 3984 } else { 3985 headLocalPTS = transformedPTS; 3986 } 3987 } 3988 3989 // adjust the head buffer's PTS to reflect the portion of the head buffer 3990 // that has already been consumed 3991 int64_t effectivePTS = headLocalPTS + 3992 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3993 3994 // Calculate the delta in samples between the head of the input buffer 3995 // queue and the start of the next output buffer that will be written. 3996 // If the transformation fails because of over or underflow, it means 3997 // that the sample's position in the output stream is so far out of 3998 // whack that it should just be dropped. 3999 int64_t sampleDelta; 4000 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4001 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4002 mTimedBufferQueue.removeAt(0); 4003 continue; 4004 } 4005 if (!mLocalTimeToSampleTransform.doForwardTransform( 4006 (effectivePTS - pts) << 32, &sampleDelta)) { 4007 ALOGV("*** too late during sample rate transform: dropped buffer"); 4008 mTimedBufferQueue.removeAt(0); 4009 continue; 4010 } 4011 4012 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4013 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4014 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4015 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4016 4017 // if the delta between the ideal placement for the next input sample and 4018 // the current output position is within this threshold, then we will 4019 // concatenate the next input samples to the previous output 4020 const int64_t kSampleContinuityThreshold = 4021 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4022 4023 // if this is the first buffer of audio that we're emitting from this track 4024 // then it should be almost exactly on time. 4025 const int64_t kSampleStartupThreshold = 1LL << 32; 4026 4027 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4028 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4029 // the next input is close enough to being on time, so concatenate it 4030 // with the last output 4031 timedYieldSamples(buffer); 4032 4033 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4034 return NO_ERROR; 4035 } else if (sampleDelta > 0) { 4036 // the gap between the current output position and the proper start of 4037 // the next input sample is too big, so fill it with silence 4038 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4039 4040 timedYieldSilence(framesUntilNextInput, buffer); 4041 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4042 return NO_ERROR; 4043 } else { 4044 // the next input sample is late 4045 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4046 size_t onTimeSamplePosition = 4047 head.position() + lateFrames * mCblk->frameSize; 4048 4049 if (onTimeSamplePosition > head.buffer()->size()) { 4050 // all the remaining samples in the head are too late, so 4051 // drop it and move on 4052 ALOGV("*** too late: dropped buffer"); 4053 mTimedBufferQueue.removeAt(0); 4054 continue; 4055 } else { 4056 // skip over the late samples 4057 head.setPosition(onTimeSamplePosition); 4058 4059 // yield the available samples 4060 timedYieldSamples(buffer); 4061 4062 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4063 return NO_ERROR; 4064 } 4065 } 4066 } 4067} 4068 4069// Yield samples from the timed buffer queue head up to the given output 4070// buffer's capacity. 4071// 4072// Caller must hold mTimedBufferQueueLock 4073void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4074 AudioBufferProvider::Buffer* buffer) { 4075 4076 const TimedBuffer& head = mTimedBufferQueue[0]; 4077 4078 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4079 head.position()); 4080 4081 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4082 mCblk->frameSize); 4083 size_t framesRequested = buffer->frameCount; 4084 buffer->frameCount = min(framesLeftInHead, framesRequested); 4085 4086 mTimedAudioOutputOnTime = true; 4087} 4088 4089// Yield samples of silence up to the given output buffer's capacity 4090// 4091// Caller must hold mTimedBufferQueueLock 4092void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4093 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4094 4095 // lazily allocate a buffer filled with silence 4096 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4097 delete [] mTimedSilenceBuffer; 4098 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4099 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4100 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4101 } 4102 4103 buffer->raw = mTimedSilenceBuffer; 4104 size_t framesRequested = buffer->frameCount; 4105 buffer->frameCount = min(numFrames, framesRequested); 4106 4107 mTimedAudioOutputOnTime = false; 4108} 4109 4110void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4111 AudioBufferProvider::Buffer* buffer) { 4112 4113 Mutex::Autolock _l(mTimedBufferQueueLock); 4114 4115 // If the buffer which was just released is part of the buffer at the head 4116 // of the queue, be sure to update the amt of the buffer which has been 4117 // consumed. If the buffer being returned is not part of the head of the 4118 // queue, its either because the buffer is part of the silence buffer, or 4119 // because the head of the timed queue was trimmed after the mixer called 4120 // getNextBuffer but before the mixer called releaseBuffer. 4121 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4122 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4123 4124 void* start = head.buffer()->pointer(); 4125 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4126 4127 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4128 head.setPosition(head.position() + 4129 (buffer->frameCount * mCblk->frameSize)); 4130 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4131 mTimedBufferQueue.removeAt(0); 4132 } 4133 } 4134 } 4135 4136 buffer->raw = 0; 4137 buffer->frameCount = 0; 4138} 4139 4140uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4141 Mutex::Autolock _l(mTimedBufferQueueLock); 4142 4143 uint32_t frames = 0; 4144 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4145 const TimedBuffer& tb = mTimedBufferQueue[i]; 4146 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4147 } 4148 4149 return frames; 4150} 4151 4152AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4153 : mPTS(0), mPosition(0) {} 4154 4155AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4156 const sp<IMemory>& buffer, int64_t pts) 4157 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4158 4159// ---------------------------------------------------------------------------- 4160 4161// RecordTrack constructor must be called with AudioFlinger::mLock held 4162AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4163 RecordThread *thread, 4164 const sp<Client>& client, 4165 uint32_t sampleRate, 4166 audio_format_t format, 4167 uint32_t channelMask, 4168 int frameCount, 4169 int sessionId) 4170 : TrackBase(thread, client, sampleRate, format, 4171 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4172 mOverflow(false) 4173{ 4174 if (mCblk != NULL) { 4175 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4176 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4177 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4178 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4179 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4180 } else { 4181 mCblk->frameSize = sizeof(int8_t); 4182 } 4183 } 4184} 4185 4186AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4187{ 4188 sp<ThreadBase> thread = mThread.promote(); 4189 if (thread != 0) { 4190 AudioSystem::releaseInput(thread->id()); 4191 } 4192} 4193 4194status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4195{ 4196 audio_track_cblk_t* cblk = this->cblk(); 4197 uint32_t framesAvail; 4198 uint32_t framesReq = buffer->frameCount; 4199 4200 // Check if last stepServer failed, try to step now 4201 if (mStepServerFailed) { 4202 if (!step()) goto getNextBuffer_exit; 4203 ALOGV("stepServer recovered"); 4204 mStepServerFailed = false; 4205 } 4206 4207 framesAvail = cblk->framesAvailable_l(); 4208 4209 if (CC_LIKELY(framesAvail)) { 4210 uint32_t s = cblk->server; 4211 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4212 4213 if (framesReq > framesAvail) { 4214 framesReq = framesAvail; 4215 } 4216 if (s + framesReq > bufferEnd) { 4217 framesReq = bufferEnd - s; 4218 } 4219 4220 buffer->raw = getBuffer(s, framesReq); 4221 if (buffer->raw == NULL) goto getNextBuffer_exit; 4222 4223 buffer->frameCount = framesReq; 4224 return NO_ERROR; 4225 } 4226 4227getNextBuffer_exit: 4228 buffer->raw = NULL; 4229 buffer->frameCount = 0; 4230 return NOT_ENOUGH_DATA; 4231} 4232 4233status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4234{ 4235 sp<ThreadBase> thread = mThread.promote(); 4236 if (thread != 0) { 4237 RecordThread *recordThread = (RecordThread *)thread.get(); 4238 return recordThread->start(this, tid); 4239 } else { 4240 return BAD_VALUE; 4241 } 4242} 4243 4244void AudioFlinger::RecordThread::RecordTrack::stop() 4245{ 4246 sp<ThreadBase> thread = mThread.promote(); 4247 if (thread != 0) { 4248 RecordThread *recordThread = (RecordThread *)thread.get(); 4249 recordThread->stop(this); 4250 TrackBase::reset(); 4251 // Force overerrun condition to avoid false overrun callback until first data is 4252 // read from buffer 4253 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4254 } 4255} 4256 4257void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4258{ 4259 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4260 (mClient == 0) ? getpid_cached : mClient->pid(), 4261 mFormat, 4262 mChannelMask, 4263 mSessionId, 4264 mFrameCount, 4265 mState, 4266 mCblk->sampleRate, 4267 mCblk->server, 4268 mCblk->user); 4269} 4270 4271 4272// ---------------------------------------------------------------------------- 4273 4274AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4275 PlaybackThread *playbackThread, 4276 DuplicatingThread *sourceThread, 4277 uint32_t sampleRate, 4278 audio_format_t format, 4279 uint32_t channelMask, 4280 int frameCount) 4281 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4282 mActive(false), mSourceThread(sourceThread) 4283{ 4284 4285 if (mCblk != NULL) { 4286 mCblk->flags |= CBLK_DIRECTION_OUT; 4287 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4288 mOutBuffer.frameCount = 0; 4289 playbackThread->mTracks.add(this); 4290 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4291 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4292 mCblk, mBuffer, mCblk->buffers, 4293 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4294 } else { 4295 ALOGW("Error creating output track on thread %p", playbackThread); 4296 } 4297} 4298 4299AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4300{ 4301 clearBufferQueue(); 4302} 4303 4304status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4305{ 4306 status_t status = Track::start(tid); 4307 if (status != NO_ERROR) { 4308 return status; 4309 } 4310 4311 mActive = true; 4312 mRetryCount = 127; 4313 return status; 4314} 4315 4316void AudioFlinger::PlaybackThread::OutputTrack::stop() 4317{ 4318 Track::stop(); 4319 clearBufferQueue(); 4320 mOutBuffer.frameCount = 0; 4321 mActive = false; 4322} 4323 4324bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4325{ 4326 Buffer *pInBuffer; 4327 Buffer inBuffer; 4328 uint32_t channelCount = mChannelCount; 4329 bool outputBufferFull = false; 4330 inBuffer.frameCount = frames; 4331 inBuffer.i16 = data; 4332 4333 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4334 4335 if (!mActive && frames != 0) { 4336 start(0); 4337 sp<ThreadBase> thread = mThread.promote(); 4338 if (thread != 0) { 4339 MixerThread *mixerThread = (MixerThread *)thread.get(); 4340 if (mCblk->frameCount > frames){ 4341 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4342 uint32_t startFrames = (mCblk->frameCount - frames); 4343 pInBuffer = new Buffer; 4344 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4345 pInBuffer->frameCount = startFrames; 4346 pInBuffer->i16 = pInBuffer->mBuffer; 4347 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4348 mBufferQueue.add(pInBuffer); 4349 } else { 4350 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4351 } 4352 } 4353 } 4354 } 4355 4356 while (waitTimeLeftMs) { 4357 // First write pending buffers, then new data 4358 if (mBufferQueue.size()) { 4359 pInBuffer = mBufferQueue.itemAt(0); 4360 } else { 4361 pInBuffer = &inBuffer; 4362 } 4363 4364 if (pInBuffer->frameCount == 0) { 4365 break; 4366 } 4367 4368 if (mOutBuffer.frameCount == 0) { 4369 mOutBuffer.frameCount = pInBuffer->frameCount; 4370 nsecs_t startTime = systemTime(); 4371 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4372 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4373 outputBufferFull = true; 4374 break; 4375 } 4376 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4377 if (waitTimeLeftMs >= waitTimeMs) { 4378 waitTimeLeftMs -= waitTimeMs; 4379 } else { 4380 waitTimeLeftMs = 0; 4381 } 4382 } 4383 4384 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4385 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4386 mCblk->stepUser(outFrames); 4387 pInBuffer->frameCount -= outFrames; 4388 pInBuffer->i16 += outFrames * channelCount; 4389 mOutBuffer.frameCount -= outFrames; 4390 mOutBuffer.i16 += outFrames * channelCount; 4391 4392 if (pInBuffer->frameCount == 0) { 4393 if (mBufferQueue.size()) { 4394 mBufferQueue.removeAt(0); 4395 delete [] pInBuffer->mBuffer; 4396 delete pInBuffer; 4397 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4398 } else { 4399 break; 4400 } 4401 } 4402 } 4403 4404 // If we could not write all frames, allocate a buffer and queue it for next time. 4405 if (inBuffer.frameCount) { 4406 sp<ThreadBase> thread = mThread.promote(); 4407 if (thread != 0 && !thread->standby()) { 4408 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4409 pInBuffer = new Buffer; 4410 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4411 pInBuffer->frameCount = inBuffer.frameCount; 4412 pInBuffer->i16 = pInBuffer->mBuffer; 4413 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4414 mBufferQueue.add(pInBuffer); 4415 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4416 } else { 4417 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4418 } 4419 } 4420 } 4421 4422 // Calling write() with a 0 length buffer, means that no more data will be written: 4423 // If no more buffers are pending, fill output track buffer to make sure it is started 4424 // by output mixer. 4425 if (frames == 0 && mBufferQueue.size() == 0) { 4426 if (mCblk->user < mCblk->frameCount) { 4427 frames = mCblk->frameCount - mCblk->user; 4428 pInBuffer = new Buffer; 4429 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4430 pInBuffer->frameCount = frames; 4431 pInBuffer->i16 = pInBuffer->mBuffer; 4432 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4433 mBufferQueue.add(pInBuffer); 4434 } else if (mActive) { 4435 stop(); 4436 } 4437 } 4438 4439 return outputBufferFull; 4440} 4441 4442status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4443{ 4444 int active; 4445 status_t result; 4446 audio_track_cblk_t* cblk = mCblk; 4447 uint32_t framesReq = buffer->frameCount; 4448 4449// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4450 buffer->frameCount = 0; 4451 4452 uint32_t framesAvail = cblk->framesAvailable(); 4453 4454 4455 if (framesAvail == 0) { 4456 Mutex::Autolock _l(cblk->lock); 4457 goto start_loop_here; 4458 while (framesAvail == 0) { 4459 active = mActive; 4460 if (CC_UNLIKELY(!active)) { 4461 ALOGV("Not active and NO_MORE_BUFFERS"); 4462 return NO_MORE_BUFFERS; 4463 } 4464 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4465 if (result != NO_ERROR) { 4466 return NO_MORE_BUFFERS; 4467 } 4468 // read the server count again 4469 start_loop_here: 4470 framesAvail = cblk->framesAvailable_l(); 4471 } 4472 } 4473 4474// if (framesAvail < framesReq) { 4475// return NO_MORE_BUFFERS; 4476// } 4477 4478 if (framesReq > framesAvail) { 4479 framesReq = framesAvail; 4480 } 4481 4482 uint32_t u = cblk->user; 4483 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4484 4485 if (u + framesReq > bufferEnd) { 4486 framesReq = bufferEnd - u; 4487 } 4488 4489 buffer->frameCount = framesReq; 4490 buffer->raw = (void *)cblk->buffer(u); 4491 return NO_ERROR; 4492} 4493 4494 4495void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4496{ 4497 size_t size = mBufferQueue.size(); 4498 4499 for (size_t i = 0; i < size; i++) { 4500 Buffer *pBuffer = mBufferQueue.itemAt(i); 4501 delete [] pBuffer->mBuffer; 4502 delete pBuffer; 4503 } 4504 mBufferQueue.clear(); 4505} 4506 4507// ---------------------------------------------------------------------------- 4508 4509AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4510 : RefBase(), 4511 mAudioFlinger(audioFlinger), 4512 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4513 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4514 mPid(pid), 4515 mTimedTrackCount(0) 4516{ 4517 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4518} 4519 4520// Client destructor must be called with AudioFlinger::mLock held 4521AudioFlinger::Client::~Client() 4522{ 4523 mAudioFlinger->removeClient_l(mPid); 4524} 4525 4526sp<MemoryDealer> AudioFlinger::Client::heap() const 4527{ 4528 return mMemoryDealer; 4529} 4530 4531// Reserve one of the limited slots for a timed audio track associated 4532// with this client 4533bool AudioFlinger::Client::reserveTimedTrack() 4534{ 4535 const int kMaxTimedTracksPerClient = 4; 4536 4537 Mutex::Autolock _l(mTimedTrackLock); 4538 4539 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4540 ALOGW("can not create timed track - pid %d has exceeded the limit", 4541 mPid); 4542 return false; 4543 } 4544 4545 mTimedTrackCount++; 4546 return true; 4547} 4548 4549// Release a slot for a timed audio track 4550void AudioFlinger::Client::releaseTimedTrack() 4551{ 4552 Mutex::Autolock _l(mTimedTrackLock); 4553 mTimedTrackCount--; 4554} 4555 4556// ---------------------------------------------------------------------------- 4557 4558AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4559 const sp<IAudioFlingerClient>& client, 4560 pid_t pid) 4561 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4562{ 4563} 4564 4565AudioFlinger::NotificationClient::~NotificationClient() 4566{ 4567} 4568 4569void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4570{ 4571 sp<NotificationClient> keep(this); 4572 mAudioFlinger->removeNotificationClient(mPid); 4573} 4574 4575// ---------------------------------------------------------------------------- 4576 4577AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4578 : BnAudioTrack(), 4579 mTrack(track) 4580{ 4581} 4582 4583AudioFlinger::TrackHandle::~TrackHandle() { 4584 // just stop the track on deletion, associated resources 4585 // will be freed from the main thread once all pending buffers have 4586 // been played. Unless it's not in the active track list, in which 4587 // case we free everything now... 4588 mTrack->destroy(); 4589} 4590 4591sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4592 return mTrack->getCblk(); 4593} 4594 4595status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4596 return mTrack->start(tid); 4597} 4598 4599void AudioFlinger::TrackHandle::stop() { 4600 mTrack->stop(); 4601} 4602 4603void AudioFlinger::TrackHandle::flush() { 4604 mTrack->flush(); 4605} 4606 4607void AudioFlinger::TrackHandle::mute(bool e) { 4608 mTrack->mute(e); 4609} 4610 4611void AudioFlinger::TrackHandle::pause() { 4612 mTrack->pause(); 4613} 4614 4615status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4616{ 4617 return mTrack->attachAuxEffect(EffectId); 4618} 4619 4620status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4621 sp<IMemory>* buffer) { 4622 if (!mTrack->isTimedTrack()) 4623 return INVALID_OPERATION; 4624 4625 PlaybackThread::TimedTrack* tt = 4626 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4627 return tt->allocateTimedBuffer(size, buffer); 4628} 4629 4630status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4631 int64_t pts) { 4632 if (!mTrack->isTimedTrack()) 4633 return INVALID_OPERATION; 4634 4635 PlaybackThread::TimedTrack* tt = 4636 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4637 return tt->queueTimedBuffer(buffer, pts); 4638} 4639 4640status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4641 const LinearTransform& xform, int target) { 4642 4643 if (!mTrack->isTimedTrack()) 4644 return INVALID_OPERATION; 4645 4646 PlaybackThread::TimedTrack* tt = 4647 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4648 return tt->setMediaTimeTransform( 4649 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4650} 4651 4652status_t AudioFlinger::TrackHandle::onTransact( 4653 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4654{ 4655 return BnAudioTrack::onTransact(code, data, reply, flags); 4656} 4657 4658// ---------------------------------------------------------------------------- 4659 4660sp<IAudioRecord> AudioFlinger::openRecord( 4661 pid_t pid, 4662 audio_io_handle_t input, 4663 uint32_t sampleRate, 4664 audio_format_t format, 4665 uint32_t channelMask, 4666 int frameCount, 4667 // FIXME dead, remove from IAudioFlinger 4668 uint32_t flags, 4669 int *sessionId, 4670 status_t *status) 4671{ 4672 sp<RecordThread::RecordTrack> recordTrack; 4673 sp<RecordHandle> recordHandle; 4674 sp<Client> client; 4675 status_t lStatus; 4676 RecordThread *thread; 4677 size_t inFrameCount; 4678 int lSessionId; 4679 4680 // check calling permissions 4681 if (!recordingAllowed()) { 4682 lStatus = PERMISSION_DENIED; 4683 goto Exit; 4684 } 4685 4686 // add client to list 4687 { // scope for mLock 4688 Mutex::Autolock _l(mLock); 4689 thread = checkRecordThread_l(input); 4690 if (thread == NULL) { 4691 lStatus = BAD_VALUE; 4692 goto Exit; 4693 } 4694 4695 client = registerPid_l(pid); 4696 4697 // If no audio session id is provided, create one here 4698 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4699 lSessionId = *sessionId; 4700 } else { 4701 lSessionId = nextUniqueId(); 4702 if (sessionId != NULL) { 4703 *sessionId = lSessionId; 4704 } 4705 } 4706 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4707 recordTrack = thread->createRecordTrack_l(client, 4708 sampleRate, 4709 format, 4710 channelMask, 4711 frameCount, 4712 lSessionId, 4713 &lStatus); 4714 } 4715 if (lStatus != NO_ERROR) { 4716 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4717 // destructor is called by the TrackBase destructor with mLock held 4718 client.clear(); 4719 recordTrack.clear(); 4720 goto Exit; 4721 } 4722 4723 // return to handle to client 4724 recordHandle = new RecordHandle(recordTrack); 4725 lStatus = NO_ERROR; 4726 4727Exit: 4728 if (status) { 4729 *status = lStatus; 4730 } 4731 return recordHandle; 4732} 4733 4734// ---------------------------------------------------------------------------- 4735 4736AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4737 : BnAudioRecord(), 4738 mRecordTrack(recordTrack) 4739{ 4740} 4741 4742AudioFlinger::RecordHandle::~RecordHandle() { 4743 stop(); 4744} 4745 4746sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4747 return mRecordTrack->getCblk(); 4748} 4749 4750status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4751 ALOGV("RecordHandle::start()"); 4752 return mRecordTrack->start(tid); 4753} 4754 4755void AudioFlinger::RecordHandle::stop() { 4756 ALOGV("RecordHandle::stop()"); 4757 mRecordTrack->stop(); 4758} 4759 4760status_t AudioFlinger::RecordHandle::onTransact( 4761 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4762{ 4763 return BnAudioRecord::onTransact(code, data, reply, flags); 4764} 4765 4766// ---------------------------------------------------------------------------- 4767 4768AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4769 AudioStreamIn *input, 4770 uint32_t sampleRate, 4771 uint32_t channels, 4772 audio_io_handle_t id, 4773 uint32_t device) : 4774 ThreadBase(audioFlinger, id, device, RECORD), 4775 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4776 // mRsmpInIndex and mInputBytes set by readInputParameters() 4777 mReqChannelCount(popcount(channels)), 4778 mReqSampleRate(sampleRate) 4779 // mBytesRead is only meaningful while active, and so is cleared in start() 4780 // (but might be better to also clear here for dump?) 4781{ 4782 snprintf(mName, kNameLength, "AudioIn_%d", id); 4783 4784 readInputParameters(); 4785} 4786 4787 4788AudioFlinger::RecordThread::~RecordThread() 4789{ 4790 delete[] mRsmpInBuffer; 4791 delete mResampler; 4792 delete[] mRsmpOutBuffer; 4793} 4794 4795void AudioFlinger::RecordThread::onFirstRef() 4796{ 4797 run(mName, PRIORITY_URGENT_AUDIO); 4798} 4799 4800status_t AudioFlinger::RecordThread::readyToRun() 4801{ 4802 status_t status = initCheck(); 4803 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4804 return status; 4805} 4806 4807bool AudioFlinger::RecordThread::threadLoop() 4808{ 4809 AudioBufferProvider::Buffer buffer; 4810 sp<RecordTrack> activeTrack; 4811 Vector< sp<EffectChain> > effectChains; 4812 4813 nsecs_t lastWarning = 0; 4814 4815 acquireWakeLock(); 4816 4817 // start recording 4818 while (!exitPending()) { 4819 4820 processConfigEvents(); 4821 4822 { // scope for mLock 4823 Mutex::Autolock _l(mLock); 4824 checkForNewParameters_l(); 4825 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4826 if (!mStandby) { 4827 mInput->stream->common.standby(&mInput->stream->common); 4828 mStandby = true; 4829 } 4830 4831 if (exitPending()) break; 4832 4833 releaseWakeLock_l(); 4834 ALOGV("RecordThread: loop stopping"); 4835 // go to sleep 4836 mWaitWorkCV.wait(mLock); 4837 ALOGV("RecordThread: loop starting"); 4838 acquireWakeLock_l(); 4839 continue; 4840 } 4841 if (mActiveTrack != 0) { 4842 if (mActiveTrack->mState == TrackBase::PAUSING) { 4843 if (!mStandby) { 4844 mInput->stream->common.standby(&mInput->stream->common); 4845 mStandby = true; 4846 } 4847 mActiveTrack.clear(); 4848 mStartStopCond.broadcast(); 4849 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4850 if (mReqChannelCount != mActiveTrack->channelCount()) { 4851 mActiveTrack.clear(); 4852 mStartStopCond.broadcast(); 4853 } else if (mBytesRead != 0) { 4854 // record start succeeds only if first read from audio input 4855 // succeeds 4856 if (mBytesRead > 0) { 4857 mActiveTrack->mState = TrackBase::ACTIVE; 4858 } else { 4859 mActiveTrack.clear(); 4860 } 4861 mStartStopCond.broadcast(); 4862 } 4863 mStandby = false; 4864 } 4865 } 4866 lockEffectChains_l(effectChains); 4867 } 4868 4869 if (mActiveTrack != 0) { 4870 if (mActiveTrack->mState != TrackBase::ACTIVE && 4871 mActiveTrack->mState != TrackBase::RESUMING) { 4872 unlockEffectChains(effectChains); 4873 usleep(kRecordThreadSleepUs); 4874 continue; 4875 } 4876 for (size_t i = 0; i < effectChains.size(); i ++) { 4877 effectChains[i]->process_l(); 4878 } 4879 4880 buffer.frameCount = mFrameCount; 4881 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4882 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4883 size_t framesOut = buffer.frameCount; 4884 if (mResampler == NULL) { 4885 // no resampling 4886 while (framesOut) { 4887 size_t framesIn = mFrameCount - mRsmpInIndex; 4888 if (framesIn) { 4889 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4890 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4891 if (framesIn > framesOut) 4892 framesIn = framesOut; 4893 mRsmpInIndex += framesIn; 4894 framesOut -= framesIn; 4895 if ((int)mChannelCount == mReqChannelCount || 4896 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4897 memcpy(dst, src, framesIn * mFrameSize); 4898 } else { 4899 int16_t *src16 = (int16_t *)src; 4900 int16_t *dst16 = (int16_t *)dst; 4901 if (mChannelCount == 1) { 4902 while (framesIn--) { 4903 *dst16++ = *src16; 4904 *dst16++ = *src16++; 4905 } 4906 } else { 4907 while (framesIn--) { 4908 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4909 src16 += 2; 4910 } 4911 } 4912 } 4913 } 4914 if (framesOut && mFrameCount == mRsmpInIndex) { 4915 if (framesOut == mFrameCount && 4916 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4917 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4918 framesOut = 0; 4919 } else { 4920 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4921 mRsmpInIndex = 0; 4922 } 4923 if (mBytesRead < 0) { 4924 ALOGE("Error reading audio input"); 4925 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4926 // Force input into standby so that it tries to 4927 // recover at next read attempt 4928 mInput->stream->common.standby(&mInput->stream->common); 4929 usleep(kRecordThreadSleepUs); 4930 } 4931 mRsmpInIndex = mFrameCount; 4932 framesOut = 0; 4933 buffer.frameCount = 0; 4934 } 4935 } 4936 } 4937 } else { 4938 // resampling 4939 4940 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4941 // alter output frame count as if we were expecting stereo samples 4942 if (mChannelCount == 1 && mReqChannelCount == 1) { 4943 framesOut >>= 1; 4944 } 4945 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4946 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4947 // are 32 bit aligned which should be always true. 4948 if (mChannelCount == 2 && mReqChannelCount == 1) { 4949 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4950 // the resampler always outputs stereo samples: do post stereo to mono conversion 4951 int16_t *src = (int16_t *)mRsmpOutBuffer; 4952 int16_t *dst = buffer.i16; 4953 while (framesOut--) { 4954 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4955 src += 2; 4956 } 4957 } else { 4958 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4959 } 4960 4961 } 4962 mActiveTrack->releaseBuffer(&buffer); 4963 mActiveTrack->overflow(); 4964 } 4965 // client isn't retrieving buffers fast enough 4966 else { 4967 if (!mActiveTrack->setOverflow()) { 4968 nsecs_t now = systemTime(); 4969 if ((now - lastWarning) > kWarningThrottleNs) { 4970 ALOGW("RecordThread: buffer overflow"); 4971 lastWarning = now; 4972 } 4973 } 4974 // Release the processor for a while before asking for a new buffer. 4975 // This will give the application more chance to read from the buffer and 4976 // clear the overflow. 4977 usleep(kRecordThreadSleepUs); 4978 } 4979 } 4980 // enable changes in effect chain 4981 unlockEffectChains(effectChains); 4982 effectChains.clear(); 4983 } 4984 4985 if (!mStandby) { 4986 mInput->stream->common.standby(&mInput->stream->common); 4987 } 4988 mActiveTrack.clear(); 4989 4990 mStartStopCond.broadcast(); 4991 4992 releaseWakeLock(); 4993 4994 ALOGV("RecordThread %p exiting", this); 4995 return false; 4996} 4997 4998 4999sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5000 const sp<AudioFlinger::Client>& client, 5001 uint32_t sampleRate, 5002 audio_format_t format, 5003 int channelMask, 5004 int frameCount, 5005 int sessionId, 5006 status_t *status) 5007{ 5008 sp<RecordTrack> track; 5009 status_t lStatus; 5010 5011 lStatus = initCheck(); 5012 if (lStatus != NO_ERROR) { 5013 ALOGE("Audio driver not initialized."); 5014 goto Exit; 5015 } 5016 5017 { // scope for mLock 5018 Mutex::Autolock _l(mLock); 5019 5020 track = new RecordTrack(this, client, sampleRate, 5021 format, channelMask, frameCount, sessionId); 5022 5023 if (track->getCblk() == 0) { 5024 lStatus = NO_MEMORY; 5025 goto Exit; 5026 } 5027 5028 mTrack = track.get(); 5029 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5030 bool suspend = audio_is_bluetooth_sco_device( 5031 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5032 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5033 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5034 } 5035 lStatus = NO_ERROR; 5036 5037Exit: 5038 if (status) { 5039 *status = lStatus; 5040 } 5041 return track; 5042} 5043 5044status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5045{ 5046 ALOGV("RecordThread::start tid=%d", tid); 5047 sp <ThreadBase> strongMe = this; 5048 status_t status = NO_ERROR; 5049 { 5050 AutoMutex lock(mLock); 5051 if (mActiveTrack != 0) { 5052 if (recordTrack != mActiveTrack.get()) { 5053 status = -EBUSY; 5054 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5055 mActiveTrack->mState = TrackBase::ACTIVE; 5056 } 5057 return status; 5058 } 5059 5060 recordTrack->mState = TrackBase::IDLE; 5061 mActiveTrack = recordTrack; 5062 mLock.unlock(); 5063 status_t status = AudioSystem::startInput(mId); 5064 mLock.lock(); 5065 if (status != NO_ERROR) { 5066 mActiveTrack.clear(); 5067 return status; 5068 } 5069 mRsmpInIndex = mFrameCount; 5070 mBytesRead = 0; 5071 if (mResampler != NULL) { 5072 mResampler->reset(); 5073 } 5074 mActiveTrack->mState = TrackBase::RESUMING; 5075 // signal thread to start 5076 ALOGV("Signal record thread"); 5077 mWaitWorkCV.signal(); 5078 // do not wait for mStartStopCond if exiting 5079 if (exitPending()) { 5080 mActiveTrack.clear(); 5081 status = INVALID_OPERATION; 5082 goto startError; 5083 } 5084 mStartStopCond.wait(mLock); 5085 if (mActiveTrack == 0) { 5086 ALOGV("Record failed to start"); 5087 status = BAD_VALUE; 5088 goto startError; 5089 } 5090 ALOGV("Record started OK"); 5091 return status; 5092 } 5093startError: 5094 AudioSystem::stopInput(mId); 5095 return status; 5096} 5097 5098void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5099 ALOGV("RecordThread::stop"); 5100 sp <ThreadBase> strongMe = this; 5101 { 5102 AutoMutex lock(mLock); 5103 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5104 mActiveTrack->mState = TrackBase::PAUSING; 5105 // do not wait for mStartStopCond if exiting 5106 if (exitPending()) { 5107 return; 5108 } 5109 mStartStopCond.wait(mLock); 5110 // if we have been restarted, recordTrack == mActiveTrack.get() here 5111 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5112 mLock.unlock(); 5113 AudioSystem::stopInput(mId); 5114 mLock.lock(); 5115 ALOGV("Record stopped OK"); 5116 } 5117 } 5118 } 5119} 5120 5121status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5122{ 5123 const size_t SIZE = 256; 5124 char buffer[SIZE]; 5125 String8 result; 5126 5127 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5128 result.append(buffer); 5129 5130 if (mActiveTrack != 0) { 5131 result.append("Active Track:\n"); 5132 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5133 mActiveTrack->dump(buffer, SIZE); 5134 result.append(buffer); 5135 5136 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5137 result.append(buffer); 5138 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5139 result.append(buffer); 5140 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5141 result.append(buffer); 5142 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5143 result.append(buffer); 5144 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5145 result.append(buffer); 5146 5147 5148 } else { 5149 result.append("No record client\n"); 5150 } 5151 write(fd, result.string(), result.size()); 5152 5153 dumpBase(fd, args); 5154 dumpEffectChains(fd, args); 5155 5156 return NO_ERROR; 5157} 5158 5159status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5160{ 5161 size_t framesReq = buffer->frameCount; 5162 size_t framesReady = mFrameCount - mRsmpInIndex; 5163 int channelCount; 5164 5165 if (framesReady == 0) { 5166 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5167 if (mBytesRead < 0) { 5168 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5169 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5170 // Force input into standby so that it tries to 5171 // recover at next read attempt 5172 mInput->stream->common.standby(&mInput->stream->common); 5173 usleep(kRecordThreadSleepUs); 5174 } 5175 buffer->raw = NULL; 5176 buffer->frameCount = 0; 5177 return NOT_ENOUGH_DATA; 5178 } 5179 mRsmpInIndex = 0; 5180 framesReady = mFrameCount; 5181 } 5182 5183 if (framesReq > framesReady) { 5184 framesReq = framesReady; 5185 } 5186 5187 if (mChannelCount == 1 && mReqChannelCount == 2) { 5188 channelCount = 1; 5189 } else { 5190 channelCount = 2; 5191 } 5192 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5193 buffer->frameCount = framesReq; 5194 return NO_ERROR; 5195} 5196 5197void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5198{ 5199 mRsmpInIndex += buffer->frameCount; 5200 buffer->frameCount = 0; 5201} 5202 5203bool AudioFlinger::RecordThread::checkForNewParameters_l() 5204{ 5205 bool reconfig = false; 5206 5207 while (!mNewParameters.isEmpty()) { 5208 status_t status = NO_ERROR; 5209 String8 keyValuePair = mNewParameters[0]; 5210 AudioParameter param = AudioParameter(keyValuePair); 5211 int value; 5212 audio_format_t reqFormat = mFormat; 5213 int reqSamplingRate = mReqSampleRate; 5214 int reqChannelCount = mReqChannelCount; 5215 5216 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5217 reqSamplingRate = value; 5218 reconfig = true; 5219 } 5220 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5221 reqFormat = (audio_format_t) value; 5222 reconfig = true; 5223 } 5224 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5225 reqChannelCount = popcount(value); 5226 reconfig = true; 5227 } 5228 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5229 // do not accept frame count changes if tracks are open as the track buffer 5230 // size depends on frame count and correct behavior would not be guaranteed 5231 // if frame count is changed after track creation 5232 if (mActiveTrack != 0) { 5233 status = INVALID_OPERATION; 5234 } else { 5235 reconfig = true; 5236 } 5237 } 5238 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5239 // forward device change to effects that have requested to be 5240 // aware of attached audio device. 5241 for (size_t i = 0; i < mEffectChains.size(); i++) { 5242 mEffectChains[i]->setDevice_l(value); 5243 } 5244 // store input device and output device but do not forward output device to audio HAL. 5245 // Note that status is ignored by the caller for output device 5246 // (see AudioFlinger::setParameters() 5247 if (value & AUDIO_DEVICE_OUT_ALL) { 5248 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5249 status = BAD_VALUE; 5250 } else { 5251 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5252 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5253 if (mTrack != NULL) { 5254 bool suspend = audio_is_bluetooth_sco_device( 5255 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5256 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5257 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5258 } 5259 } 5260 mDevice |= (uint32_t)value; 5261 } 5262 if (status == NO_ERROR) { 5263 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5264 if (status == INVALID_OPERATION) { 5265 mInput->stream->common.standby(&mInput->stream->common); 5266 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5267 } 5268 if (reconfig) { 5269 if (status == BAD_VALUE && 5270 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5271 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5272 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5273 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5274 (reqChannelCount < 3)) { 5275 status = NO_ERROR; 5276 } 5277 if (status == NO_ERROR) { 5278 readInputParameters(); 5279 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5280 } 5281 } 5282 } 5283 5284 mNewParameters.removeAt(0); 5285 5286 mParamStatus = status; 5287 mParamCond.signal(); 5288 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5289 // already timed out waiting for the status and will never signal the condition. 5290 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5291 } 5292 return reconfig; 5293} 5294 5295String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5296{ 5297 char *s; 5298 String8 out_s8 = String8(); 5299 5300 Mutex::Autolock _l(mLock); 5301 if (initCheck() != NO_ERROR) { 5302 return out_s8; 5303 } 5304 5305 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5306 out_s8 = String8(s); 5307 free(s); 5308 return out_s8; 5309} 5310 5311void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5312 AudioSystem::OutputDescriptor desc; 5313 void *param2 = NULL; 5314 5315 switch (event) { 5316 case AudioSystem::INPUT_OPENED: 5317 case AudioSystem::INPUT_CONFIG_CHANGED: 5318 desc.channels = mChannelMask; 5319 desc.samplingRate = mSampleRate; 5320 desc.format = mFormat; 5321 desc.frameCount = mFrameCount; 5322 desc.latency = 0; 5323 param2 = &desc; 5324 break; 5325 5326 case AudioSystem::INPUT_CLOSED: 5327 default: 5328 break; 5329 } 5330 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5331} 5332 5333void AudioFlinger::RecordThread::readInputParameters() 5334{ 5335 delete mRsmpInBuffer; 5336 // mRsmpInBuffer is always assigned a new[] below 5337 delete mRsmpOutBuffer; 5338 mRsmpOutBuffer = NULL; 5339 delete mResampler; 5340 mResampler = NULL; 5341 5342 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5343 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5344 mChannelCount = (uint16_t)popcount(mChannelMask); 5345 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5346 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5347 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5348 mFrameCount = mInputBytes / mFrameSize; 5349 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5350 5351 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5352 { 5353 int channelCount; 5354 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5355 // stereo to mono post process as the resampler always outputs stereo. 5356 if (mChannelCount == 1 && mReqChannelCount == 2) { 5357 channelCount = 1; 5358 } else { 5359 channelCount = 2; 5360 } 5361 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5362 mResampler->setSampleRate(mSampleRate); 5363 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5364 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5365 5366 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5367 if (mChannelCount == 1 && mReqChannelCount == 1) { 5368 mFrameCount >>= 1; 5369 } 5370 5371 } 5372 mRsmpInIndex = mFrameCount; 5373} 5374 5375unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5376{ 5377 Mutex::Autolock _l(mLock); 5378 if (initCheck() != NO_ERROR) { 5379 return 0; 5380 } 5381 5382 return mInput->stream->get_input_frames_lost(mInput->stream); 5383} 5384 5385uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5386{ 5387 Mutex::Autolock _l(mLock); 5388 uint32_t result = 0; 5389 if (getEffectChain_l(sessionId) != 0) { 5390 result = EFFECT_SESSION; 5391 } 5392 5393 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5394 result |= TRACK_SESSION; 5395 } 5396 5397 return result; 5398} 5399 5400AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5401{ 5402 Mutex::Autolock _l(mLock); 5403 return mTrack; 5404} 5405 5406AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5407{ 5408 Mutex::Autolock _l(mLock); 5409 return mInput; 5410} 5411 5412AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5413{ 5414 Mutex::Autolock _l(mLock); 5415 AudioStreamIn *input = mInput; 5416 mInput = NULL; 5417 return input; 5418} 5419 5420// this method must always be called either with ThreadBase mLock held or inside the thread loop 5421audio_stream_t* AudioFlinger::RecordThread::stream() 5422{ 5423 if (mInput == NULL) { 5424 return NULL; 5425 } 5426 return &mInput->stream->common; 5427} 5428 5429 5430// ---------------------------------------------------------------------------- 5431 5432audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5433 uint32_t *pSamplingRate, 5434 audio_format_t *pFormat, 5435 uint32_t *pChannels, 5436 uint32_t *pLatencyMs, 5437 uint32_t flags) 5438{ 5439 status_t status; 5440 PlaybackThread *thread = NULL; 5441 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5442 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5443 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5444 uint32_t channels = pChannels ? *pChannels : 0; 5445 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5446 audio_stream_out_t *outStream; 5447 audio_hw_device_t *outHwDev; 5448 5449 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5450 pDevices ? *pDevices : 0, 5451 samplingRate, 5452 format, 5453 channels, 5454 flags); 5455 5456 if (pDevices == NULL || *pDevices == 0) { 5457 return 0; 5458 } 5459 5460 Mutex::Autolock _l(mLock); 5461 5462 outHwDev = findSuitableHwDev_l(*pDevices); 5463 if (outHwDev == NULL) 5464 return 0; 5465 5466 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5467 &channels, &samplingRate, &outStream); 5468 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5469 outStream, 5470 samplingRate, 5471 format, 5472 channels, 5473 status); 5474 5475 mHardwareStatus = AUDIO_HW_IDLE; 5476 if (outStream != NULL) { 5477 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5478 audio_io_handle_t id = nextUniqueId(); 5479 5480 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5481 (format != AUDIO_FORMAT_PCM_16_BIT) || 5482 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5483 thread = new DirectOutputThread(this, output, id, *pDevices); 5484 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5485 } else { 5486 thread = new MixerThread(this, output, id, *pDevices); 5487 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5488 } 5489 mPlaybackThreads.add(id, thread); 5490 5491 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5492 if (pFormat != NULL) *pFormat = format; 5493 if (pChannels != NULL) *pChannels = channels; 5494 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5495 5496 // notify client processes of the new output creation 5497 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5498 return id; 5499 } 5500 5501 return 0; 5502} 5503 5504audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5505 audio_io_handle_t output2) 5506{ 5507 Mutex::Autolock _l(mLock); 5508 MixerThread *thread1 = checkMixerThread_l(output1); 5509 MixerThread *thread2 = checkMixerThread_l(output2); 5510 5511 if (thread1 == NULL || thread2 == NULL) { 5512 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5513 return 0; 5514 } 5515 5516 audio_io_handle_t id = nextUniqueId(); 5517 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5518 thread->addOutputTrack(thread2); 5519 mPlaybackThreads.add(id, thread); 5520 // notify client processes of the new output creation 5521 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5522 return id; 5523} 5524 5525status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5526{ 5527 // keep strong reference on the playback thread so that 5528 // it is not destroyed while exit() is executed 5529 sp <PlaybackThread> thread; 5530 { 5531 Mutex::Autolock _l(mLock); 5532 thread = checkPlaybackThread_l(output); 5533 if (thread == NULL) { 5534 return BAD_VALUE; 5535 } 5536 5537 ALOGV("closeOutput() %d", output); 5538 5539 if (thread->type() == ThreadBase::MIXER) { 5540 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5541 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5542 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5543 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5544 } 5545 } 5546 } 5547 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5548 mPlaybackThreads.removeItem(output); 5549 } 5550 thread->exit(); 5551 // The thread entity (active unit of execution) is no longer running here, 5552 // but the ThreadBase container still exists. 5553 5554 if (thread->type() != ThreadBase::DUPLICATING) { 5555 AudioStreamOut *out = thread->clearOutput(); 5556 assert(out != NULL); 5557 // from now on thread->mOutput is NULL 5558 out->hwDev->close_output_stream(out->hwDev, out->stream); 5559 delete out; 5560 } 5561 return NO_ERROR; 5562} 5563 5564status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5565{ 5566 Mutex::Autolock _l(mLock); 5567 PlaybackThread *thread = checkPlaybackThread_l(output); 5568 5569 if (thread == NULL) { 5570 return BAD_VALUE; 5571 } 5572 5573 ALOGV("suspendOutput() %d", output); 5574 thread->suspend(); 5575 5576 return NO_ERROR; 5577} 5578 5579status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5580{ 5581 Mutex::Autolock _l(mLock); 5582 PlaybackThread *thread = checkPlaybackThread_l(output); 5583 5584 if (thread == NULL) { 5585 return BAD_VALUE; 5586 } 5587 5588 ALOGV("restoreOutput() %d", output); 5589 5590 thread->restore(); 5591 5592 return NO_ERROR; 5593} 5594 5595audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5596 uint32_t *pSamplingRate, 5597 audio_format_t *pFormat, 5598 uint32_t *pChannels, 5599 audio_in_acoustics_t acoustics) 5600{ 5601 status_t status; 5602 RecordThread *thread = NULL; 5603 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5604 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5605 uint32_t channels = pChannels ? *pChannels : 0; 5606 uint32_t reqSamplingRate = samplingRate; 5607 audio_format_t reqFormat = format; 5608 uint32_t reqChannels = channels; 5609 audio_stream_in_t *inStream; 5610 audio_hw_device_t *inHwDev; 5611 5612 if (pDevices == NULL || *pDevices == 0) { 5613 return 0; 5614 } 5615 5616 Mutex::Autolock _l(mLock); 5617 5618 inHwDev = findSuitableHwDev_l(*pDevices); 5619 if (inHwDev == NULL) 5620 return 0; 5621 5622 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5623 &channels, &samplingRate, 5624 acoustics, 5625 &inStream); 5626 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5627 inStream, 5628 samplingRate, 5629 format, 5630 channels, 5631 acoustics, 5632 status); 5633 5634 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5635 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5636 // or stereo to mono conversions on 16 bit PCM inputs. 5637 if (inStream == NULL && status == BAD_VALUE && 5638 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5639 (samplingRate <= 2 * reqSamplingRate) && 5640 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5641 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5642 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5643 &channels, &samplingRate, 5644 acoustics, 5645 &inStream); 5646 } 5647 5648 if (inStream != NULL) { 5649 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5650 5651 audio_io_handle_t id = nextUniqueId(); 5652 // Start record thread 5653 // RecorThread require both input and output device indication to forward to audio 5654 // pre processing modules 5655 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5656 thread = new RecordThread(this, 5657 input, 5658 reqSamplingRate, 5659 reqChannels, 5660 id, 5661 device); 5662 mRecordThreads.add(id, thread); 5663 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5664 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5665 if (pFormat != NULL) *pFormat = format; 5666 if (pChannels != NULL) *pChannels = reqChannels; 5667 5668 input->stream->common.standby(&input->stream->common); 5669 5670 // notify client processes of the new input creation 5671 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5672 return id; 5673 } 5674 5675 return 0; 5676} 5677 5678status_t AudioFlinger::closeInput(audio_io_handle_t input) 5679{ 5680 // keep strong reference on the record thread so that 5681 // it is not destroyed while exit() is executed 5682 sp <RecordThread> thread; 5683 { 5684 Mutex::Autolock _l(mLock); 5685 thread = checkRecordThread_l(input); 5686 if (thread == NULL) { 5687 return BAD_VALUE; 5688 } 5689 5690 ALOGV("closeInput() %d", input); 5691 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5692 mRecordThreads.removeItem(input); 5693 } 5694 thread->exit(); 5695 // The thread entity (active unit of execution) is no longer running here, 5696 // but the ThreadBase container still exists. 5697 5698 AudioStreamIn *in = thread->clearInput(); 5699 assert(in != NULL); 5700 // from now on thread->mInput is NULL 5701 in->hwDev->close_input_stream(in->hwDev, in->stream); 5702 delete in; 5703 5704 return NO_ERROR; 5705} 5706 5707status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5708{ 5709 Mutex::Autolock _l(mLock); 5710 MixerThread *dstThread = checkMixerThread_l(output); 5711 if (dstThread == NULL) { 5712 ALOGW("setStreamOutput() bad output id %d", output); 5713 return BAD_VALUE; 5714 } 5715 5716 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5717 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5718 5719 dstThread->setStreamValid(stream, true); 5720 5721 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5722 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5723 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5724 MixerThread *srcThread = (MixerThread *)thread; 5725 srcThread->setStreamValid(stream, false); 5726 srcThread->invalidateTracks(stream); 5727 } 5728 } 5729 5730 return NO_ERROR; 5731} 5732 5733 5734int AudioFlinger::newAudioSessionId() 5735{ 5736 return nextUniqueId(); 5737} 5738 5739void AudioFlinger::acquireAudioSessionId(int audioSession) 5740{ 5741 Mutex::Autolock _l(mLock); 5742 pid_t caller = IPCThreadState::self()->getCallingPid(); 5743 ALOGV("acquiring %d from %d", audioSession, caller); 5744 size_t num = mAudioSessionRefs.size(); 5745 for (size_t i = 0; i< num; i++) { 5746 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5747 if (ref->sessionid == audioSession && ref->pid == caller) { 5748 ref->cnt++; 5749 ALOGV(" incremented refcount to %d", ref->cnt); 5750 return; 5751 } 5752 } 5753 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5754 ALOGV(" added new entry for %d", audioSession); 5755} 5756 5757void AudioFlinger::releaseAudioSessionId(int audioSession) 5758{ 5759 Mutex::Autolock _l(mLock); 5760 pid_t caller = IPCThreadState::self()->getCallingPid(); 5761 ALOGV("releasing %d from %d", audioSession, caller); 5762 size_t num = mAudioSessionRefs.size(); 5763 for (size_t i = 0; i< num; i++) { 5764 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5765 if (ref->sessionid == audioSession && ref->pid == caller) { 5766 ref->cnt--; 5767 ALOGV(" decremented refcount to %d", ref->cnt); 5768 if (ref->cnt == 0) { 5769 mAudioSessionRefs.removeAt(i); 5770 delete ref; 5771 purgeStaleEffects_l(); 5772 } 5773 return; 5774 } 5775 } 5776 ALOGW("session id %d not found for pid %d", audioSession, caller); 5777} 5778 5779void AudioFlinger::purgeStaleEffects_l() { 5780 5781 ALOGV("purging stale effects"); 5782 5783 Vector< sp<EffectChain> > chains; 5784 5785 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5786 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5787 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5788 sp<EffectChain> ec = t->mEffectChains[j]; 5789 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5790 chains.push(ec); 5791 } 5792 } 5793 } 5794 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5795 sp<RecordThread> t = mRecordThreads.valueAt(i); 5796 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5797 sp<EffectChain> ec = t->mEffectChains[j]; 5798 chains.push(ec); 5799 } 5800 } 5801 5802 for (size_t i = 0; i < chains.size(); i++) { 5803 sp<EffectChain> ec = chains[i]; 5804 int sessionid = ec->sessionId(); 5805 sp<ThreadBase> t = ec->mThread.promote(); 5806 if (t == 0) { 5807 continue; 5808 } 5809 size_t numsessionrefs = mAudioSessionRefs.size(); 5810 bool found = false; 5811 for (size_t k = 0; k < numsessionrefs; k++) { 5812 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5813 if (ref->sessionid == sessionid) { 5814 ALOGV(" session %d still exists for %d with %d refs", 5815 sessionid, ref->pid, ref->cnt); 5816 found = true; 5817 break; 5818 } 5819 } 5820 if (!found) { 5821 // remove all effects from the chain 5822 while (ec->mEffects.size()) { 5823 sp<EffectModule> effect = ec->mEffects[0]; 5824 effect->unPin(); 5825 Mutex::Autolock _l (t->mLock); 5826 t->removeEffect_l(effect); 5827 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5828 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5829 if (handle != 0) { 5830 handle->mEffect.clear(); 5831 if (handle->mHasControl && handle->mEnabled) { 5832 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5833 } 5834 } 5835 } 5836 AudioSystem::unregisterEffect(effect->id()); 5837 } 5838 } 5839 } 5840 return; 5841} 5842 5843// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5844AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5845{ 5846 return mPlaybackThreads.valueFor(output).get(); 5847} 5848 5849// checkMixerThread_l() must be called with AudioFlinger::mLock held 5850AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5851{ 5852 PlaybackThread *thread = checkPlaybackThread_l(output); 5853 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5854} 5855 5856// checkRecordThread_l() must be called with AudioFlinger::mLock held 5857AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5858{ 5859 return mRecordThreads.valueFor(input).get(); 5860} 5861 5862uint32_t AudioFlinger::nextUniqueId() 5863{ 5864 return android_atomic_inc(&mNextUniqueId); 5865} 5866 5867AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5868{ 5869 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5870 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5871 AudioStreamOut *output = thread->getOutput(); 5872 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5873 return thread; 5874 } 5875 } 5876 return NULL; 5877} 5878 5879uint32_t AudioFlinger::primaryOutputDevice_l() 5880{ 5881 PlaybackThread *thread = primaryPlaybackThread_l(); 5882 5883 if (thread == NULL) { 5884 return 0; 5885 } 5886 5887 return thread->device(); 5888} 5889 5890 5891// ---------------------------------------------------------------------------- 5892// Effect management 5893// ---------------------------------------------------------------------------- 5894 5895 5896status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5897{ 5898 Mutex::Autolock _l(mLock); 5899 return EffectQueryNumberEffects(numEffects); 5900} 5901 5902status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5903{ 5904 Mutex::Autolock _l(mLock); 5905 return EffectQueryEffect(index, descriptor); 5906} 5907 5908status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5909 effect_descriptor_t *descriptor) const 5910{ 5911 Mutex::Autolock _l(mLock); 5912 return EffectGetDescriptor(pUuid, descriptor); 5913} 5914 5915 5916sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5917 effect_descriptor_t *pDesc, 5918 const sp<IEffectClient>& effectClient, 5919 int32_t priority, 5920 audio_io_handle_t io, 5921 int sessionId, 5922 status_t *status, 5923 int *id, 5924 int *enabled) 5925{ 5926 status_t lStatus = NO_ERROR; 5927 sp<EffectHandle> handle; 5928 effect_descriptor_t desc; 5929 5930 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5931 pid, effectClient.get(), priority, sessionId, io); 5932 5933 if (pDesc == NULL) { 5934 lStatus = BAD_VALUE; 5935 goto Exit; 5936 } 5937 5938 // check audio settings permission for global effects 5939 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5940 lStatus = PERMISSION_DENIED; 5941 goto Exit; 5942 } 5943 5944 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5945 // that can only be created by audio policy manager (running in same process) 5946 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5947 lStatus = PERMISSION_DENIED; 5948 goto Exit; 5949 } 5950 5951 if (io == 0) { 5952 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5953 // output must be specified by AudioPolicyManager when using session 5954 // AUDIO_SESSION_OUTPUT_STAGE 5955 lStatus = BAD_VALUE; 5956 goto Exit; 5957 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5958 // if the output returned by getOutputForEffect() is removed before we lock the 5959 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5960 // and we will exit safely 5961 io = AudioSystem::getOutputForEffect(&desc); 5962 } 5963 } 5964 5965 { 5966 Mutex::Autolock _l(mLock); 5967 5968 5969 if (!EffectIsNullUuid(&pDesc->uuid)) { 5970 // if uuid is specified, request effect descriptor 5971 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5972 if (lStatus < 0) { 5973 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5974 goto Exit; 5975 } 5976 } else { 5977 // if uuid is not specified, look for an available implementation 5978 // of the required type in effect factory 5979 if (EffectIsNullUuid(&pDesc->type)) { 5980 ALOGW("createEffect() no effect type"); 5981 lStatus = BAD_VALUE; 5982 goto Exit; 5983 } 5984 uint32_t numEffects = 0; 5985 effect_descriptor_t d; 5986 d.flags = 0; // prevent compiler warning 5987 bool found = false; 5988 5989 lStatus = EffectQueryNumberEffects(&numEffects); 5990 if (lStatus < 0) { 5991 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5992 goto Exit; 5993 } 5994 for (uint32_t i = 0; i < numEffects; i++) { 5995 lStatus = EffectQueryEffect(i, &desc); 5996 if (lStatus < 0) { 5997 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5998 continue; 5999 } 6000 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6001 // If matching type found save effect descriptor. If the session is 6002 // 0 and the effect is not auxiliary, continue enumeration in case 6003 // an auxiliary version of this effect type is available 6004 found = true; 6005 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6006 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6007 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6008 break; 6009 } 6010 } 6011 } 6012 if (!found) { 6013 lStatus = BAD_VALUE; 6014 ALOGW("createEffect() effect not found"); 6015 goto Exit; 6016 } 6017 // For same effect type, chose auxiliary version over insert version if 6018 // connect to output mix (Compliance to OpenSL ES) 6019 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6020 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6021 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6022 } 6023 } 6024 6025 // Do not allow auxiliary effects on a session different from 0 (output mix) 6026 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6027 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6028 lStatus = INVALID_OPERATION; 6029 goto Exit; 6030 } 6031 6032 // check recording permission for visualizer 6033 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6034 !recordingAllowed()) { 6035 lStatus = PERMISSION_DENIED; 6036 goto Exit; 6037 } 6038 6039 // return effect descriptor 6040 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6041 6042 // If output is not specified try to find a matching audio session ID in one of the 6043 // output threads. 6044 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6045 // because of code checking output when entering the function. 6046 // Note: io is never 0 when creating an effect on an input 6047 if (io == 0) { 6048 // look for the thread where the specified audio session is present 6049 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6050 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6051 io = mPlaybackThreads.keyAt(i); 6052 break; 6053 } 6054 } 6055 if (io == 0) { 6056 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6057 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6058 io = mRecordThreads.keyAt(i); 6059 break; 6060 } 6061 } 6062 } 6063 // If no output thread contains the requested session ID, default to 6064 // first output. The effect chain will be moved to the correct output 6065 // thread when a track with the same session ID is created 6066 if (io == 0 && mPlaybackThreads.size()) { 6067 io = mPlaybackThreads.keyAt(0); 6068 } 6069 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6070 } 6071 ThreadBase *thread = checkRecordThread_l(io); 6072 if (thread == NULL) { 6073 thread = checkPlaybackThread_l(io); 6074 if (thread == NULL) { 6075 ALOGE("createEffect() unknown output thread"); 6076 lStatus = BAD_VALUE; 6077 goto Exit; 6078 } 6079 } 6080 6081 sp<Client> client = registerPid_l(pid); 6082 6083 // create effect on selected output thread 6084 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6085 &desc, enabled, &lStatus); 6086 if (handle != 0 && id != NULL) { 6087 *id = handle->id(); 6088 } 6089 } 6090 6091Exit: 6092 if(status) { 6093 *status = lStatus; 6094 } 6095 return handle; 6096} 6097 6098status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6099 audio_io_handle_t dstOutput) 6100{ 6101 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6102 sessionId, srcOutput, dstOutput); 6103 Mutex::Autolock _l(mLock); 6104 if (srcOutput == dstOutput) { 6105 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6106 return NO_ERROR; 6107 } 6108 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6109 if (srcThread == NULL) { 6110 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6111 return BAD_VALUE; 6112 } 6113 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6114 if (dstThread == NULL) { 6115 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6116 return BAD_VALUE; 6117 } 6118 6119 Mutex::Autolock _dl(dstThread->mLock); 6120 Mutex::Autolock _sl(srcThread->mLock); 6121 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6122 6123 return NO_ERROR; 6124} 6125 6126// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6127status_t AudioFlinger::moveEffectChain_l(int sessionId, 6128 AudioFlinger::PlaybackThread *srcThread, 6129 AudioFlinger::PlaybackThread *dstThread, 6130 bool reRegister) 6131{ 6132 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6133 sessionId, srcThread, dstThread); 6134 6135 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6136 if (chain == 0) { 6137 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6138 sessionId, srcThread); 6139 return INVALID_OPERATION; 6140 } 6141 6142 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6143 // so that a new chain is created with correct parameters when first effect is added. This is 6144 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6145 // removed. 6146 srcThread->removeEffectChain_l(chain); 6147 6148 // transfer all effects one by one so that new effect chain is created on new thread with 6149 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6150 audio_io_handle_t dstOutput = dstThread->id(); 6151 sp<EffectChain> dstChain; 6152 uint32_t strategy = 0; // prevent compiler warning 6153 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6154 while (effect != 0) { 6155 srcThread->removeEffect_l(effect); 6156 dstThread->addEffect_l(effect); 6157 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6158 if (effect->state() == EffectModule::ACTIVE || 6159 effect->state() == EffectModule::STOPPING) { 6160 effect->start(); 6161 } 6162 // if the move request is not received from audio policy manager, the effect must be 6163 // re-registered with the new strategy and output 6164 if (dstChain == 0) { 6165 dstChain = effect->chain().promote(); 6166 if (dstChain == 0) { 6167 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6168 srcThread->addEffect_l(effect); 6169 return NO_INIT; 6170 } 6171 strategy = dstChain->strategy(); 6172 } 6173 if (reRegister) { 6174 AudioSystem::unregisterEffect(effect->id()); 6175 AudioSystem::registerEffect(&effect->desc(), 6176 dstOutput, 6177 strategy, 6178 sessionId, 6179 effect->id()); 6180 } 6181 effect = chain->getEffectFromId_l(0); 6182 } 6183 6184 return NO_ERROR; 6185} 6186 6187 6188// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6189sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6190 const sp<AudioFlinger::Client>& client, 6191 const sp<IEffectClient>& effectClient, 6192 int32_t priority, 6193 int sessionId, 6194 effect_descriptor_t *desc, 6195 int *enabled, 6196 status_t *status 6197 ) 6198{ 6199 sp<EffectModule> effect; 6200 sp<EffectHandle> handle; 6201 status_t lStatus; 6202 sp<EffectChain> chain; 6203 bool chainCreated = false; 6204 bool effectCreated = false; 6205 bool effectRegistered = false; 6206 6207 lStatus = initCheck(); 6208 if (lStatus != NO_ERROR) { 6209 ALOGW("createEffect_l() Audio driver not initialized."); 6210 goto Exit; 6211 } 6212 6213 // Do not allow effects with session ID 0 on direct output or duplicating threads 6214 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6215 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6216 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6217 desc->name, sessionId); 6218 lStatus = BAD_VALUE; 6219 goto Exit; 6220 } 6221 // Only Pre processor effects are allowed on input threads and only on input threads 6222 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6223 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6224 desc->name, desc->flags, mType); 6225 lStatus = BAD_VALUE; 6226 goto Exit; 6227 } 6228 6229 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6230 6231 { // scope for mLock 6232 Mutex::Autolock _l(mLock); 6233 6234 // check for existing effect chain with the requested audio session 6235 chain = getEffectChain_l(sessionId); 6236 if (chain == 0) { 6237 // create a new chain for this session 6238 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6239 chain = new EffectChain(this, sessionId); 6240 addEffectChain_l(chain); 6241 chain->setStrategy(getStrategyForSession_l(sessionId)); 6242 chainCreated = true; 6243 } else { 6244 effect = chain->getEffectFromDesc_l(desc); 6245 } 6246 6247 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6248 6249 if (effect == 0) { 6250 int id = mAudioFlinger->nextUniqueId(); 6251 // Check CPU and memory usage 6252 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6253 if (lStatus != NO_ERROR) { 6254 goto Exit; 6255 } 6256 effectRegistered = true; 6257 // create a new effect module if none present in the chain 6258 effect = new EffectModule(this, chain, desc, id, sessionId); 6259 lStatus = effect->status(); 6260 if (lStatus != NO_ERROR) { 6261 goto Exit; 6262 } 6263 lStatus = chain->addEffect_l(effect); 6264 if (lStatus != NO_ERROR) { 6265 goto Exit; 6266 } 6267 effectCreated = true; 6268 6269 effect->setDevice(mDevice); 6270 effect->setMode(mAudioFlinger->getMode()); 6271 } 6272 // create effect handle and connect it to effect module 6273 handle = new EffectHandle(effect, client, effectClient, priority); 6274 lStatus = effect->addHandle(handle); 6275 if (enabled != NULL) { 6276 *enabled = (int)effect->isEnabled(); 6277 } 6278 } 6279 6280Exit: 6281 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6282 Mutex::Autolock _l(mLock); 6283 if (effectCreated) { 6284 chain->removeEffect_l(effect); 6285 } 6286 if (effectRegistered) { 6287 AudioSystem::unregisterEffect(effect->id()); 6288 } 6289 if (chainCreated) { 6290 removeEffectChain_l(chain); 6291 } 6292 handle.clear(); 6293 } 6294 6295 if(status) { 6296 *status = lStatus; 6297 } 6298 return handle; 6299} 6300 6301sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6302{ 6303 sp<EffectChain> chain = getEffectChain_l(sessionId); 6304 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6305} 6306 6307// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6308// PlaybackThread::mLock held 6309status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6310{ 6311 // check for existing effect chain with the requested audio session 6312 int sessionId = effect->sessionId(); 6313 sp<EffectChain> chain = getEffectChain_l(sessionId); 6314 bool chainCreated = false; 6315 6316 if (chain == 0) { 6317 // create a new chain for this session 6318 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6319 chain = new EffectChain(this, sessionId); 6320 addEffectChain_l(chain); 6321 chain->setStrategy(getStrategyForSession_l(sessionId)); 6322 chainCreated = true; 6323 } 6324 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6325 6326 if (chain->getEffectFromId_l(effect->id()) != 0) { 6327 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6328 this, effect->desc().name, chain.get()); 6329 return BAD_VALUE; 6330 } 6331 6332 status_t status = chain->addEffect_l(effect); 6333 if (status != NO_ERROR) { 6334 if (chainCreated) { 6335 removeEffectChain_l(chain); 6336 } 6337 return status; 6338 } 6339 6340 effect->setDevice(mDevice); 6341 effect->setMode(mAudioFlinger->getMode()); 6342 return NO_ERROR; 6343} 6344 6345void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6346 6347 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6348 effect_descriptor_t desc = effect->desc(); 6349 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6350 detachAuxEffect_l(effect->id()); 6351 } 6352 6353 sp<EffectChain> chain = effect->chain().promote(); 6354 if (chain != 0) { 6355 // remove effect chain if removing last effect 6356 if (chain->removeEffect_l(effect) == 0) { 6357 removeEffectChain_l(chain); 6358 } 6359 } else { 6360 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6361 } 6362} 6363 6364void AudioFlinger::ThreadBase::lockEffectChains_l( 6365 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6366{ 6367 effectChains = mEffectChains; 6368 for (size_t i = 0; i < mEffectChains.size(); i++) { 6369 mEffectChains[i]->lock(); 6370 } 6371} 6372 6373void AudioFlinger::ThreadBase::unlockEffectChains( 6374 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6375{ 6376 for (size_t i = 0; i < effectChains.size(); i++) { 6377 effectChains[i]->unlock(); 6378 } 6379} 6380 6381sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6382{ 6383 Mutex::Autolock _l(mLock); 6384 return getEffectChain_l(sessionId); 6385} 6386 6387sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6388{ 6389 size_t size = mEffectChains.size(); 6390 for (size_t i = 0; i < size; i++) { 6391 if (mEffectChains[i]->sessionId() == sessionId) { 6392 return mEffectChains[i]; 6393 } 6394 } 6395 return 0; 6396} 6397 6398void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6399{ 6400 Mutex::Autolock _l(mLock); 6401 size_t size = mEffectChains.size(); 6402 for (size_t i = 0; i < size; i++) { 6403 mEffectChains[i]->setMode_l(mode); 6404 } 6405} 6406 6407void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6408 const wp<EffectHandle>& handle, 6409 bool unpinIfLast) { 6410 6411 Mutex::Autolock _l(mLock); 6412 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6413 // delete the effect module if removing last handle on it 6414 if (effect->removeHandle(handle) == 0) { 6415 if (!effect->isPinned() || unpinIfLast) { 6416 removeEffect_l(effect); 6417 AudioSystem::unregisterEffect(effect->id()); 6418 } 6419 } 6420} 6421 6422status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6423{ 6424 int session = chain->sessionId(); 6425 int16_t *buffer = mMixBuffer; 6426 bool ownsBuffer = false; 6427 6428 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6429 if (session > 0) { 6430 // Only one effect chain can be present in direct output thread and it uses 6431 // the mix buffer as input 6432 if (mType != DIRECT) { 6433 size_t numSamples = mFrameCount * mChannelCount; 6434 buffer = new int16_t[numSamples]; 6435 memset(buffer, 0, numSamples * sizeof(int16_t)); 6436 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6437 ownsBuffer = true; 6438 } 6439 6440 // Attach all tracks with same session ID to this chain. 6441 for (size_t i = 0; i < mTracks.size(); ++i) { 6442 sp<Track> track = mTracks[i]; 6443 if (session == track->sessionId()) { 6444 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6445 track->setMainBuffer(buffer); 6446 chain->incTrackCnt(); 6447 } 6448 } 6449 6450 // indicate all active tracks in the chain 6451 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6452 sp<Track> track = mActiveTracks[i].promote(); 6453 if (track == 0) continue; 6454 if (session == track->sessionId()) { 6455 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6456 chain->incActiveTrackCnt(); 6457 } 6458 } 6459 } 6460 6461 chain->setInBuffer(buffer, ownsBuffer); 6462 chain->setOutBuffer(mMixBuffer); 6463 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6464 // chains list in order to be processed last as it contains output stage effects 6465 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6466 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6467 // after track specific effects and before output stage 6468 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6469 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6470 // Effect chain for other sessions are inserted at beginning of effect 6471 // chains list to be processed before output mix effects. Relative order between other 6472 // sessions is not important 6473 size_t size = mEffectChains.size(); 6474 size_t i = 0; 6475 for (i = 0; i < size; i++) { 6476 if (mEffectChains[i]->sessionId() < session) break; 6477 } 6478 mEffectChains.insertAt(chain, i); 6479 checkSuspendOnAddEffectChain_l(chain); 6480 6481 return NO_ERROR; 6482} 6483 6484size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6485{ 6486 int session = chain->sessionId(); 6487 6488 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6489 6490 for (size_t i = 0; i < mEffectChains.size(); i++) { 6491 if (chain == mEffectChains[i]) { 6492 mEffectChains.removeAt(i); 6493 // detach all active tracks from the chain 6494 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6495 sp<Track> track = mActiveTracks[i].promote(); 6496 if (track == 0) continue; 6497 if (session == track->sessionId()) { 6498 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6499 chain.get(), session); 6500 chain->decActiveTrackCnt(); 6501 } 6502 } 6503 6504 // detach all tracks with same session ID from this chain 6505 for (size_t i = 0; i < mTracks.size(); ++i) { 6506 sp<Track> track = mTracks[i]; 6507 if (session == track->sessionId()) { 6508 track->setMainBuffer(mMixBuffer); 6509 chain->decTrackCnt(); 6510 } 6511 } 6512 break; 6513 } 6514 } 6515 return mEffectChains.size(); 6516} 6517 6518status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6519 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6520{ 6521 Mutex::Autolock _l(mLock); 6522 return attachAuxEffect_l(track, EffectId); 6523} 6524 6525status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6526 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6527{ 6528 status_t status = NO_ERROR; 6529 6530 if (EffectId == 0) { 6531 track->setAuxBuffer(0, NULL); 6532 } else { 6533 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6534 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6535 if (effect != 0) { 6536 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6537 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6538 } else { 6539 status = INVALID_OPERATION; 6540 } 6541 } else { 6542 status = BAD_VALUE; 6543 } 6544 } 6545 return status; 6546} 6547 6548void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6549{ 6550 for (size_t i = 0; i < mTracks.size(); ++i) { 6551 sp<Track> track = mTracks[i]; 6552 if (track->auxEffectId() == effectId) { 6553 attachAuxEffect_l(track, 0); 6554 } 6555 } 6556} 6557 6558status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6559{ 6560 // only one chain per input thread 6561 if (mEffectChains.size() != 0) { 6562 return INVALID_OPERATION; 6563 } 6564 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6565 6566 chain->setInBuffer(NULL); 6567 chain->setOutBuffer(NULL); 6568 6569 checkSuspendOnAddEffectChain_l(chain); 6570 6571 mEffectChains.add(chain); 6572 6573 return NO_ERROR; 6574} 6575 6576size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6577{ 6578 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6579 ALOGW_IF(mEffectChains.size() != 1, 6580 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6581 chain.get(), mEffectChains.size(), this); 6582 if (mEffectChains.size() == 1) { 6583 mEffectChains.removeAt(0); 6584 } 6585 return 0; 6586} 6587 6588// ---------------------------------------------------------------------------- 6589// EffectModule implementation 6590// ---------------------------------------------------------------------------- 6591 6592#undef LOG_TAG 6593#define LOG_TAG "AudioFlinger::EffectModule" 6594 6595AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6596 const wp<AudioFlinger::EffectChain>& chain, 6597 effect_descriptor_t *desc, 6598 int id, 6599 int sessionId) 6600 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6601 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6602{ 6603 ALOGV("Constructor %p", this); 6604 int lStatus; 6605 if (thread == NULL) { 6606 return; 6607 } 6608 6609 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6610 6611 // create effect engine from effect factory 6612 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6613 6614 if (mStatus != NO_ERROR) { 6615 return; 6616 } 6617 lStatus = init(); 6618 if (lStatus < 0) { 6619 mStatus = lStatus; 6620 goto Error; 6621 } 6622 6623 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6624 mPinned = true; 6625 } 6626 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6627 return; 6628Error: 6629 EffectRelease(mEffectInterface); 6630 mEffectInterface = NULL; 6631 ALOGV("Constructor Error %d", mStatus); 6632} 6633 6634AudioFlinger::EffectModule::~EffectModule() 6635{ 6636 ALOGV("Destructor %p", this); 6637 if (mEffectInterface != NULL) { 6638 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6639 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6640 sp<ThreadBase> thread = mThread.promote(); 6641 if (thread != 0) { 6642 audio_stream_t *stream = thread->stream(); 6643 if (stream != NULL) { 6644 stream->remove_audio_effect(stream, mEffectInterface); 6645 } 6646 } 6647 } 6648 // release effect engine 6649 EffectRelease(mEffectInterface); 6650 } 6651} 6652 6653status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6654{ 6655 status_t status; 6656 6657 Mutex::Autolock _l(mLock); 6658 int priority = handle->priority(); 6659 size_t size = mHandles.size(); 6660 sp<EffectHandle> h; 6661 size_t i; 6662 for (i = 0; i < size; i++) { 6663 h = mHandles[i].promote(); 6664 if (h == 0) continue; 6665 if (h->priority() <= priority) break; 6666 } 6667 // if inserted in first place, move effect control from previous owner to this handle 6668 if (i == 0) { 6669 bool enabled = false; 6670 if (h != 0) { 6671 enabled = h->enabled(); 6672 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6673 } 6674 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6675 status = NO_ERROR; 6676 } else { 6677 status = ALREADY_EXISTS; 6678 } 6679 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6680 mHandles.insertAt(handle, i); 6681 return status; 6682} 6683 6684size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6685{ 6686 Mutex::Autolock _l(mLock); 6687 size_t size = mHandles.size(); 6688 size_t i; 6689 for (i = 0; i < size; i++) { 6690 if (mHandles[i] == handle) break; 6691 } 6692 if (i == size) { 6693 return size; 6694 } 6695 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6696 6697 bool enabled = false; 6698 EffectHandle *hdl = handle.unsafe_get(); 6699 if (hdl != NULL) { 6700 ALOGV("removeHandle() unsafe_get OK"); 6701 enabled = hdl->enabled(); 6702 } 6703 mHandles.removeAt(i); 6704 size = mHandles.size(); 6705 // if removed from first place, move effect control from this handle to next in line 6706 if (i == 0 && size != 0) { 6707 sp<EffectHandle> h = mHandles[0].promote(); 6708 if (h != 0) { 6709 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6710 } 6711 } 6712 6713 // Prevent calls to process() and other functions on effect interface from now on. 6714 // The effect engine will be released by the destructor when the last strong reference on 6715 // this object is released which can happen after next process is called. 6716 if (size == 0 && !mPinned) { 6717 mState = DESTROYED; 6718 } 6719 6720 return size; 6721} 6722 6723sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6724{ 6725 Mutex::Autolock _l(mLock); 6726 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6727} 6728 6729void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6730{ 6731 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6732 // keep a strong reference on this EffectModule to avoid calling the 6733 // destructor before we exit 6734 sp<EffectModule> keep(this); 6735 { 6736 sp<ThreadBase> thread = mThread.promote(); 6737 if (thread != 0) { 6738 thread->disconnectEffect(keep, handle, unpinIfLast); 6739 } 6740 } 6741} 6742 6743void AudioFlinger::EffectModule::updateState() { 6744 Mutex::Autolock _l(mLock); 6745 6746 switch (mState) { 6747 case RESTART: 6748 reset_l(); 6749 // FALL THROUGH 6750 6751 case STARTING: 6752 // clear auxiliary effect input buffer for next accumulation 6753 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6754 memset(mConfig.inputCfg.buffer.raw, 6755 0, 6756 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6757 } 6758 start_l(); 6759 mState = ACTIVE; 6760 break; 6761 case STOPPING: 6762 stop_l(); 6763 mDisableWaitCnt = mMaxDisableWaitCnt; 6764 mState = STOPPED; 6765 break; 6766 case STOPPED: 6767 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6768 // turn off sequence. 6769 if (--mDisableWaitCnt == 0) { 6770 reset_l(); 6771 mState = IDLE; 6772 } 6773 break; 6774 default: //IDLE , ACTIVE, DESTROYED 6775 break; 6776 } 6777} 6778 6779void AudioFlinger::EffectModule::process() 6780{ 6781 Mutex::Autolock _l(mLock); 6782 6783 if (mState == DESTROYED || mEffectInterface == NULL || 6784 mConfig.inputCfg.buffer.raw == NULL || 6785 mConfig.outputCfg.buffer.raw == NULL) { 6786 return; 6787 } 6788 6789 if (isProcessEnabled()) { 6790 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6791 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6792 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6793 mConfig.inputCfg.buffer.s32, 6794 mConfig.inputCfg.buffer.frameCount/2); 6795 } 6796 6797 // do the actual processing in the effect engine 6798 int ret = (*mEffectInterface)->process(mEffectInterface, 6799 &mConfig.inputCfg.buffer, 6800 &mConfig.outputCfg.buffer); 6801 6802 // force transition to IDLE state when engine is ready 6803 if (mState == STOPPED && ret == -ENODATA) { 6804 mDisableWaitCnt = 1; 6805 } 6806 6807 // clear auxiliary effect input buffer for next accumulation 6808 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6809 memset(mConfig.inputCfg.buffer.raw, 0, 6810 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6811 } 6812 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6813 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6814 // If an insert effect is idle and input buffer is different from output buffer, 6815 // accumulate input onto output 6816 sp<EffectChain> chain = mChain.promote(); 6817 if (chain != 0 && chain->activeTrackCnt() != 0) { 6818 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6819 int16_t *in = mConfig.inputCfg.buffer.s16; 6820 int16_t *out = mConfig.outputCfg.buffer.s16; 6821 for (size_t i = 0; i < frameCnt; i++) { 6822 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6823 } 6824 } 6825 } 6826} 6827 6828void AudioFlinger::EffectModule::reset_l() 6829{ 6830 if (mEffectInterface == NULL) { 6831 return; 6832 } 6833 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6834} 6835 6836status_t AudioFlinger::EffectModule::configure() 6837{ 6838 uint32_t channels; 6839 if (mEffectInterface == NULL) { 6840 return NO_INIT; 6841 } 6842 6843 sp<ThreadBase> thread = mThread.promote(); 6844 if (thread == 0) { 6845 return DEAD_OBJECT; 6846 } 6847 6848 // TODO: handle configuration of effects replacing track process 6849 if (thread->channelCount() == 1) { 6850 channels = AUDIO_CHANNEL_OUT_MONO; 6851 } else { 6852 channels = AUDIO_CHANNEL_OUT_STEREO; 6853 } 6854 6855 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6856 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6857 } else { 6858 mConfig.inputCfg.channels = channels; 6859 } 6860 mConfig.outputCfg.channels = channels; 6861 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6862 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6863 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6864 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6865 mConfig.inputCfg.bufferProvider.cookie = NULL; 6866 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6867 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6868 mConfig.outputCfg.bufferProvider.cookie = NULL; 6869 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6870 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6871 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6872 // Insert effect: 6873 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6874 // always overwrites output buffer: input buffer == output buffer 6875 // - in other sessions: 6876 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6877 // other effect: overwrites output buffer: input buffer == output buffer 6878 // Auxiliary effect: 6879 // accumulates in output buffer: input buffer != output buffer 6880 // Therefore: accumulate <=> input buffer != output buffer 6881 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6882 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6883 } else { 6884 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6885 } 6886 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6887 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6888 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6889 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6890 6891 ALOGV("configure() %p thread %p buffer %p framecount %d", 6892 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6893 6894 status_t cmdStatus; 6895 uint32_t size = sizeof(int); 6896 status_t status = (*mEffectInterface)->command(mEffectInterface, 6897 EFFECT_CMD_SET_CONFIG, 6898 sizeof(effect_config_t), 6899 &mConfig, 6900 &size, 6901 &cmdStatus); 6902 if (status == 0) { 6903 status = cmdStatus; 6904 } 6905 6906 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6907 (1000 * mConfig.outputCfg.buffer.frameCount); 6908 6909 return status; 6910} 6911 6912status_t AudioFlinger::EffectModule::init() 6913{ 6914 Mutex::Autolock _l(mLock); 6915 if (mEffectInterface == NULL) { 6916 return NO_INIT; 6917 } 6918 status_t cmdStatus; 6919 uint32_t size = sizeof(status_t); 6920 status_t status = (*mEffectInterface)->command(mEffectInterface, 6921 EFFECT_CMD_INIT, 6922 0, 6923 NULL, 6924 &size, 6925 &cmdStatus); 6926 if (status == 0) { 6927 status = cmdStatus; 6928 } 6929 return status; 6930} 6931 6932status_t AudioFlinger::EffectModule::start() 6933{ 6934 Mutex::Autolock _l(mLock); 6935 return start_l(); 6936} 6937 6938status_t AudioFlinger::EffectModule::start_l() 6939{ 6940 if (mEffectInterface == NULL) { 6941 return NO_INIT; 6942 } 6943 status_t cmdStatus; 6944 uint32_t size = sizeof(status_t); 6945 status_t status = (*mEffectInterface)->command(mEffectInterface, 6946 EFFECT_CMD_ENABLE, 6947 0, 6948 NULL, 6949 &size, 6950 &cmdStatus); 6951 if (status == 0) { 6952 status = cmdStatus; 6953 } 6954 if (status == 0 && 6955 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6956 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6957 sp<ThreadBase> thread = mThread.promote(); 6958 if (thread != 0) { 6959 audio_stream_t *stream = thread->stream(); 6960 if (stream != NULL) { 6961 stream->add_audio_effect(stream, mEffectInterface); 6962 } 6963 } 6964 } 6965 return status; 6966} 6967 6968status_t AudioFlinger::EffectModule::stop() 6969{ 6970 Mutex::Autolock _l(mLock); 6971 return stop_l(); 6972} 6973 6974status_t AudioFlinger::EffectModule::stop_l() 6975{ 6976 if (mEffectInterface == NULL) { 6977 return NO_INIT; 6978 } 6979 status_t cmdStatus; 6980 uint32_t size = sizeof(status_t); 6981 status_t status = (*mEffectInterface)->command(mEffectInterface, 6982 EFFECT_CMD_DISABLE, 6983 0, 6984 NULL, 6985 &size, 6986 &cmdStatus); 6987 if (status == 0) { 6988 status = cmdStatus; 6989 } 6990 if (status == 0 && 6991 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6992 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6993 sp<ThreadBase> thread = mThread.promote(); 6994 if (thread != 0) { 6995 audio_stream_t *stream = thread->stream(); 6996 if (stream != NULL) { 6997 stream->remove_audio_effect(stream, mEffectInterface); 6998 } 6999 } 7000 } 7001 return status; 7002} 7003 7004status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7005 uint32_t cmdSize, 7006 void *pCmdData, 7007 uint32_t *replySize, 7008 void *pReplyData) 7009{ 7010 Mutex::Autolock _l(mLock); 7011// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7012 7013 if (mState == DESTROYED || mEffectInterface == NULL) { 7014 return NO_INIT; 7015 } 7016 status_t status = (*mEffectInterface)->command(mEffectInterface, 7017 cmdCode, 7018 cmdSize, 7019 pCmdData, 7020 replySize, 7021 pReplyData); 7022 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7023 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7024 for (size_t i = 1; i < mHandles.size(); i++) { 7025 sp<EffectHandle> h = mHandles[i].promote(); 7026 if (h != 0) { 7027 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7028 } 7029 } 7030 } 7031 return status; 7032} 7033 7034status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7035{ 7036 7037 Mutex::Autolock _l(mLock); 7038 ALOGV("setEnabled %p enabled %d", this, enabled); 7039 7040 if (enabled != isEnabled()) { 7041 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7042 if (enabled && status != NO_ERROR) { 7043 return status; 7044 } 7045 7046 switch (mState) { 7047 // going from disabled to enabled 7048 case IDLE: 7049 mState = STARTING; 7050 break; 7051 case STOPPED: 7052 mState = RESTART; 7053 break; 7054 case STOPPING: 7055 mState = ACTIVE; 7056 break; 7057 7058 // going from enabled to disabled 7059 case RESTART: 7060 mState = STOPPED; 7061 break; 7062 case STARTING: 7063 mState = IDLE; 7064 break; 7065 case ACTIVE: 7066 mState = STOPPING; 7067 break; 7068 case DESTROYED: 7069 return NO_ERROR; // simply ignore as we are being destroyed 7070 } 7071 for (size_t i = 1; i < mHandles.size(); i++) { 7072 sp<EffectHandle> h = mHandles[i].promote(); 7073 if (h != 0) { 7074 h->setEnabled(enabled); 7075 } 7076 } 7077 } 7078 return NO_ERROR; 7079} 7080 7081bool AudioFlinger::EffectModule::isEnabled() const 7082{ 7083 switch (mState) { 7084 case RESTART: 7085 case STARTING: 7086 case ACTIVE: 7087 return true; 7088 case IDLE: 7089 case STOPPING: 7090 case STOPPED: 7091 case DESTROYED: 7092 default: 7093 return false; 7094 } 7095} 7096 7097bool AudioFlinger::EffectModule::isProcessEnabled() const 7098{ 7099 switch (mState) { 7100 case RESTART: 7101 case ACTIVE: 7102 case STOPPING: 7103 case STOPPED: 7104 return true; 7105 case IDLE: 7106 case STARTING: 7107 case DESTROYED: 7108 default: 7109 return false; 7110 } 7111} 7112 7113status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7114{ 7115 Mutex::Autolock _l(mLock); 7116 status_t status = NO_ERROR; 7117 7118 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7119 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7120 if (isProcessEnabled() && 7121 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7122 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7123 status_t cmdStatus; 7124 uint32_t volume[2]; 7125 uint32_t *pVolume = NULL; 7126 uint32_t size = sizeof(volume); 7127 volume[0] = *left; 7128 volume[1] = *right; 7129 if (controller) { 7130 pVolume = volume; 7131 } 7132 status = (*mEffectInterface)->command(mEffectInterface, 7133 EFFECT_CMD_SET_VOLUME, 7134 size, 7135 volume, 7136 &size, 7137 pVolume); 7138 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7139 *left = volume[0]; 7140 *right = volume[1]; 7141 } 7142 } 7143 return status; 7144} 7145 7146status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7147{ 7148 Mutex::Autolock _l(mLock); 7149 status_t status = NO_ERROR; 7150 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7151 // audio pre processing modules on RecordThread can receive both output and 7152 // input device indication in the same call 7153 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7154 if (dev) { 7155 status_t cmdStatus; 7156 uint32_t size = sizeof(status_t); 7157 7158 status = (*mEffectInterface)->command(mEffectInterface, 7159 EFFECT_CMD_SET_DEVICE, 7160 sizeof(uint32_t), 7161 &dev, 7162 &size, 7163 &cmdStatus); 7164 if (status == NO_ERROR) { 7165 status = cmdStatus; 7166 } 7167 } 7168 dev = device & AUDIO_DEVICE_IN_ALL; 7169 if (dev) { 7170 status_t cmdStatus; 7171 uint32_t size = sizeof(status_t); 7172 7173 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7174 EFFECT_CMD_SET_INPUT_DEVICE, 7175 sizeof(uint32_t), 7176 &dev, 7177 &size, 7178 &cmdStatus); 7179 if (status2 == NO_ERROR) { 7180 status2 = cmdStatus; 7181 } 7182 if (status == NO_ERROR) { 7183 status = status2; 7184 } 7185 } 7186 } 7187 return status; 7188} 7189 7190status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7191{ 7192 Mutex::Autolock _l(mLock); 7193 status_t status = NO_ERROR; 7194 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7195 status_t cmdStatus; 7196 uint32_t size = sizeof(status_t); 7197 status = (*mEffectInterface)->command(mEffectInterface, 7198 EFFECT_CMD_SET_AUDIO_MODE, 7199 sizeof(audio_mode_t), 7200 &mode, 7201 &size, 7202 &cmdStatus); 7203 if (status == NO_ERROR) { 7204 status = cmdStatus; 7205 } 7206 } 7207 return status; 7208} 7209 7210void AudioFlinger::EffectModule::setSuspended(bool suspended) 7211{ 7212 Mutex::Autolock _l(mLock); 7213 mSuspended = suspended; 7214} 7215 7216bool AudioFlinger::EffectModule::suspended() const 7217{ 7218 Mutex::Autolock _l(mLock); 7219 return mSuspended; 7220} 7221 7222status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7223{ 7224 const size_t SIZE = 256; 7225 char buffer[SIZE]; 7226 String8 result; 7227 7228 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7229 result.append(buffer); 7230 7231 bool locked = tryLock(mLock); 7232 // failed to lock - AudioFlinger is probably deadlocked 7233 if (!locked) { 7234 result.append("\t\tCould not lock Fx mutex:\n"); 7235 } 7236 7237 result.append("\t\tSession Status State Engine:\n"); 7238 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7239 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7240 result.append(buffer); 7241 7242 result.append("\t\tDescriptor:\n"); 7243 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7244 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7245 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7246 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7247 result.append(buffer); 7248 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7249 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7250 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7251 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7252 result.append(buffer); 7253 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7254 mDescriptor.apiVersion, 7255 mDescriptor.flags); 7256 result.append(buffer); 7257 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7258 mDescriptor.name); 7259 result.append(buffer); 7260 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7261 mDescriptor.implementor); 7262 result.append(buffer); 7263 7264 result.append("\t\t- Input configuration:\n"); 7265 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7266 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7267 (uint32_t)mConfig.inputCfg.buffer.raw, 7268 mConfig.inputCfg.buffer.frameCount, 7269 mConfig.inputCfg.samplingRate, 7270 mConfig.inputCfg.channels, 7271 mConfig.inputCfg.format); 7272 result.append(buffer); 7273 7274 result.append("\t\t- Output configuration:\n"); 7275 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7276 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7277 (uint32_t)mConfig.outputCfg.buffer.raw, 7278 mConfig.outputCfg.buffer.frameCount, 7279 mConfig.outputCfg.samplingRate, 7280 mConfig.outputCfg.channels, 7281 mConfig.outputCfg.format); 7282 result.append(buffer); 7283 7284 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7285 result.append(buffer); 7286 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7287 for (size_t i = 0; i < mHandles.size(); ++i) { 7288 sp<EffectHandle> handle = mHandles[i].promote(); 7289 if (handle != 0) { 7290 handle->dump(buffer, SIZE); 7291 result.append(buffer); 7292 } 7293 } 7294 7295 result.append("\n"); 7296 7297 write(fd, result.string(), result.length()); 7298 7299 if (locked) { 7300 mLock.unlock(); 7301 } 7302 7303 return NO_ERROR; 7304} 7305 7306// ---------------------------------------------------------------------------- 7307// EffectHandle implementation 7308// ---------------------------------------------------------------------------- 7309 7310#undef LOG_TAG 7311#define LOG_TAG "AudioFlinger::EffectHandle" 7312 7313AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7314 const sp<AudioFlinger::Client>& client, 7315 const sp<IEffectClient>& effectClient, 7316 int32_t priority) 7317 : BnEffect(), 7318 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7319 mPriority(priority), mHasControl(false), mEnabled(false) 7320{ 7321 ALOGV("constructor %p", this); 7322 7323 if (client == 0) { 7324 return; 7325 } 7326 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7327 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7328 if (mCblkMemory != 0) { 7329 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7330 7331 if (mCblk != NULL) { 7332 new(mCblk) effect_param_cblk_t(); 7333 mBuffer = (uint8_t *)mCblk + bufOffset; 7334 } 7335 } else { 7336 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7337 return; 7338 } 7339} 7340 7341AudioFlinger::EffectHandle::~EffectHandle() 7342{ 7343 ALOGV("Destructor %p", this); 7344 disconnect(false); 7345 ALOGV("Destructor DONE %p", this); 7346} 7347 7348status_t AudioFlinger::EffectHandle::enable() 7349{ 7350 ALOGV("enable %p", this); 7351 if (!mHasControl) return INVALID_OPERATION; 7352 if (mEffect == 0) return DEAD_OBJECT; 7353 7354 if (mEnabled) { 7355 return NO_ERROR; 7356 } 7357 7358 mEnabled = true; 7359 7360 sp<ThreadBase> thread = mEffect->thread().promote(); 7361 if (thread != 0) { 7362 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7363 } 7364 7365 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7366 if (mEffect->suspended()) { 7367 return NO_ERROR; 7368 } 7369 7370 status_t status = mEffect->setEnabled(true); 7371 if (status != NO_ERROR) { 7372 if (thread != 0) { 7373 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7374 } 7375 mEnabled = false; 7376 } 7377 return status; 7378} 7379 7380status_t AudioFlinger::EffectHandle::disable() 7381{ 7382 ALOGV("disable %p", this); 7383 if (!mHasControl) return INVALID_OPERATION; 7384 if (mEffect == 0) return DEAD_OBJECT; 7385 7386 if (!mEnabled) { 7387 return NO_ERROR; 7388 } 7389 mEnabled = false; 7390 7391 if (mEffect->suspended()) { 7392 return NO_ERROR; 7393 } 7394 7395 status_t status = mEffect->setEnabled(false); 7396 7397 sp<ThreadBase> thread = mEffect->thread().promote(); 7398 if (thread != 0) { 7399 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7400 } 7401 7402 return status; 7403} 7404 7405void AudioFlinger::EffectHandle::disconnect() 7406{ 7407 disconnect(true); 7408} 7409 7410void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7411{ 7412 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7413 if (mEffect == 0) { 7414 return; 7415 } 7416 mEffect->disconnect(this, unpinIfLast); 7417 7418 if (mHasControl && mEnabled) { 7419 sp<ThreadBase> thread = mEffect->thread().promote(); 7420 if (thread != 0) { 7421 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7422 } 7423 } 7424 7425 // release sp on module => module destructor can be called now 7426 mEffect.clear(); 7427 if (mClient != 0) { 7428 if (mCblk != NULL) { 7429 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7430 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7431 } 7432 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7433 // Client destructor must run with AudioFlinger mutex locked 7434 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7435 mClient.clear(); 7436 } 7437} 7438 7439status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7440 uint32_t cmdSize, 7441 void *pCmdData, 7442 uint32_t *replySize, 7443 void *pReplyData) 7444{ 7445// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7446// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7447 7448 // only get parameter command is permitted for applications not controlling the effect 7449 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7450 return INVALID_OPERATION; 7451 } 7452 if (mEffect == 0) return DEAD_OBJECT; 7453 if (mClient == 0) return INVALID_OPERATION; 7454 7455 // handle commands that are not forwarded transparently to effect engine 7456 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7457 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7458 // no risk to block the whole media server process or mixer threads is we are stuck here 7459 Mutex::Autolock _l(mCblk->lock); 7460 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7461 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7462 mCblk->serverIndex = 0; 7463 mCblk->clientIndex = 0; 7464 return BAD_VALUE; 7465 } 7466 status_t status = NO_ERROR; 7467 while (mCblk->serverIndex < mCblk->clientIndex) { 7468 int reply; 7469 uint32_t rsize = sizeof(int); 7470 int *p = (int *)(mBuffer + mCblk->serverIndex); 7471 int size = *p++; 7472 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7473 ALOGW("command(): invalid parameter block size"); 7474 break; 7475 } 7476 effect_param_t *param = (effect_param_t *)p; 7477 if (param->psize == 0 || param->vsize == 0) { 7478 ALOGW("command(): null parameter or value size"); 7479 mCblk->serverIndex += size; 7480 continue; 7481 } 7482 uint32_t psize = sizeof(effect_param_t) + 7483 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7484 param->vsize; 7485 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7486 psize, 7487 p, 7488 &rsize, 7489 &reply); 7490 // stop at first error encountered 7491 if (ret != NO_ERROR) { 7492 status = ret; 7493 *(int *)pReplyData = reply; 7494 break; 7495 } else if (reply != NO_ERROR) { 7496 *(int *)pReplyData = reply; 7497 break; 7498 } 7499 mCblk->serverIndex += size; 7500 } 7501 mCblk->serverIndex = 0; 7502 mCblk->clientIndex = 0; 7503 return status; 7504 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7505 *(int *)pReplyData = NO_ERROR; 7506 return enable(); 7507 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7508 *(int *)pReplyData = NO_ERROR; 7509 return disable(); 7510 } 7511 7512 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7513} 7514 7515void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7516{ 7517 ALOGV("setControl %p control %d", this, hasControl); 7518 7519 mHasControl = hasControl; 7520 mEnabled = enabled; 7521 7522 if (signal && mEffectClient != 0) { 7523 mEffectClient->controlStatusChanged(hasControl); 7524 } 7525} 7526 7527void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7528 uint32_t cmdSize, 7529 void *pCmdData, 7530 uint32_t replySize, 7531 void *pReplyData) 7532{ 7533 if (mEffectClient != 0) { 7534 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7535 } 7536} 7537 7538 7539 7540void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7541{ 7542 if (mEffectClient != 0) { 7543 mEffectClient->enableStatusChanged(enabled); 7544 } 7545} 7546 7547status_t AudioFlinger::EffectHandle::onTransact( 7548 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7549{ 7550 return BnEffect::onTransact(code, data, reply, flags); 7551} 7552 7553 7554void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7555{ 7556 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7557 7558 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7559 (mClient == 0) ? getpid_cached : mClient->pid(), 7560 mPriority, 7561 mHasControl, 7562 !locked, 7563 mCblk ? mCblk->clientIndex : 0, 7564 mCblk ? mCblk->serverIndex : 0 7565 ); 7566 7567 if (locked) { 7568 mCblk->lock.unlock(); 7569 } 7570} 7571 7572#undef LOG_TAG 7573#define LOG_TAG "AudioFlinger::EffectChain" 7574 7575AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7576 int sessionId) 7577 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7578 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7579 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7580{ 7581 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7582 if (thread == NULL) { 7583 return; 7584 } 7585 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7586 thread->frameCount(); 7587} 7588 7589AudioFlinger::EffectChain::~EffectChain() 7590{ 7591 if (mOwnInBuffer) { 7592 delete mInBuffer; 7593 } 7594 7595} 7596 7597// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7598sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7599{ 7600 size_t size = mEffects.size(); 7601 7602 for (size_t i = 0; i < size; i++) { 7603 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7604 return mEffects[i]; 7605 } 7606 } 7607 return 0; 7608} 7609 7610// getEffectFromId_l() must be called with ThreadBase::mLock held 7611sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7612{ 7613 size_t size = mEffects.size(); 7614 7615 for (size_t i = 0; i < size; i++) { 7616 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7617 if (id == 0 || mEffects[i]->id() == id) { 7618 return mEffects[i]; 7619 } 7620 } 7621 return 0; 7622} 7623 7624// getEffectFromType_l() must be called with ThreadBase::mLock held 7625sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7626 const effect_uuid_t *type) 7627{ 7628 size_t size = mEffects.size(); 7629 7630 for (size_t i = 0; i < size; i++) { 7631 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7632 return mEffects[i]; 7633 } 7634 } 7635 return 0; 7636} 7637 7638// Must be called with EffectChain::mLock locked 7639void AudioFlinger::EffectChain::process_l() 7640{ 7641 sp<ThreadBase> thread = mThread.promote(); 7642 if (thread == 0) { 7643 ALOGW("process_l(): cannot promote mixer thread"); 7644 return; 7645 } 7646 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7647 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7648 // always process effects unless no more tracks are on the session and the effect tail 7649 // has been rendered 7650 bool doProcess = true; 7651 if (!isGlobalSession) { 7652 bool tracksOnSession = (trackCnt() != 0); 7653 7654 if (!tracksOnSession && mTailBufferCount == 0) { 7655 doProcess = false; 7656 } 7657 7658 if (activeTrackCnt() == 0) { 7659 // if no track is active and the effect tail has not been rendered, 7660 // the input buffer must be cleared here as the mixer process will not do it 7661 if (tracksOnSession || mTailBufferCount > 0) { 7662 size_t numSamples = thread->frameCount() * thread->channelCount(); 7663 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7664 if (mTailBufferCount > 0) { 7665 mTailBufferCount--; 7666 } 7667 } 7668 } 7669 } 7670 7671 size_t size = mEffects.size(); 7672 if (doProcess) { 7673 for (size_t i = 0; i < size; i++) { 7674 mEffects[i]->process(); 7675 } 7676 } 7677 for (size_t i = 0; i < size; i++) { 7678 mEffects[i]->updateState(); 7679 } 7680} 7681 7682// addEffect_l() must be called with PlaybackThread::mLock held 7683status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7684{ 7685 effect_descriptor_t desc = effect->desc(); 7686 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7687 7688 Mutex::Autolock _l(mLock); 7689 effect->setChain(this); 7690 sp<ThreadBase> thread = mThread.promote(); 7691 if (thread == 0) { 7692 return NO_INIT; 7693 } 7694 effect->setThread(thread); 7695 7696 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7697 // Auxiliary effects are inserted at the beginning of mEffects vector as 7698 // they are processed first and accumulated in chain input buffer 7699 mEffects.insertAt(effect, 0); 7700 7701 // the input buffer for auxiliary effect contains mono samples in 7702 // 32 bit format. This is to avoid saturation in AudoMixer 7703 // accumulation stage. Saturation is done in EffectModule::process() before 7704 // calling the process in effect engine 7705 size_t numSamples = thread->frameCount(); 7706 int32_t *buffer = new int32_t[numSamples]; 7707 memset(buffer, 0, numSamples * sizeof(int32_t)); 7708 effect->setInBuffer((int16_t *)buffer); 7709 // auxiliary effects output samples to chain input buffer for further processing 7710 // by insert effects 7711 effect->setOutBuffer(mInBuffer); 7712 } else { 7713 // Insert effects are inserted at the end of mEffects vector as they are processed 7714 // after track and auxiliary effects. 7715 // Insert effect order as a function of indicated preference: 7716 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7717 // another effect is present 7718 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7719 // last effect claiming first position 7720 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7721 // first effect claiming last position 7722 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7723 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7724 // already present 7725 7726 size_t size = mEffects.size(); 7727 size_t idx_insert = size; 7728 ssize_t idx_insert_first = -1; 7729 ssize_t idx_insert_last = -1; 7730 7731 for (size_t i = 0; i < size; i++) { 7732 effect_descriptor_t d = mEffects[i]->desc(); 7733 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7734 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7735 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7736 // check invalid effect chaining combinations 7737 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7738 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7739 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7740 return INVALID_OPERATION; 7741 } 7742 // remember position of first insert effect and by default 7743 // select this as insert position for new effect 7744 if (idx_insert == size) { 7745 idx_insert = i; 7746 } 7747 // remember position of last insert effect claiming 7748 // first position 7749 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7750 idx_insert_first = i; 7751 } 7752 // remember position of first insert effect claiming 7753 // last position 7754 if (iPref == EFFECT_FLAG_INSERT_LAST && 7755 idx_insert_last == -1) { 7756 idx_insert_last = i; 7757 } 7758 } 7759 } 7760 7761 // modify idx_insert from first position if needed 7762 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7763 if (idx_insert_last != -1) { 7764 idx_insert = idx_insert_last; 7765 } else { 7766 idx_insert = size; 7767 } 7768 } else { 7769 if (idx_insert_first != -1) { 7770 idx_insert = idx_insert_first + 1; 7771 } 7772 } 7773 7774 // always read samples from chain input buffer 7775 effect->setInBuffer(mInBuffer); 7776 7777 // if last effect in the chain, output samples to chain 7778 // output buffer, otherwise to chain input buffer 7779 if (idx_insert == size) { 7780 if (idx_insert != 0) { 7781 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7782 mEffects[idx_insert-1]->configure(); 7783 } 7784 effect->setOutBuffer(mOutBuffer); 7785 } else { 7786 effect->setOutBuffer(mInBuffer); 7787 } 7788 mEffects.insertAt(effect, idx_insert); 7789 7790 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7791 } 7792 effect->configure(); 7793 return NO_ERROR; 7794} 7795 7796// removeEffect_l() must be called with PlaybackThread::mLock held 7797size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7798{ 7799 Mutex::Autolock _l(mLock); 7800 size_t size = mEffects.size(); 7801 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7802 7803 for (size_t i = 0; i < size; i++) { 7804 if (effect == mEffects[i]) { 7805 // calling stop here will remove pre-processing effect from the audio HAL. 7806 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7807 // the middle of a read from audio HAL 7808 if (mEffects[i]->state() == EffectModule::ACTIVE || 7809 mEffects[i]->state() == EffectModule::STOPPING) { 7810 mEffects[i]->stop(); 7811 } 7812 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7813 delete[] effect->inBuffer(); 7814 } else { 7815 if (i == size - 1 && i != 0) { 7816 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7817 mEffects[i - 1]->configure(); 7818 } 7819 } 7820 mEffects.removeAt(i); 7821 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7822 break; 7823 } 7824 } 7825 7826 return mEffects.size(); 7827} 7828 7829// setDevice_l() must be called with PlaybackThread::mLock held 7830void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7831{ 7832 size_t size = mEffects.size(); 7833 for (size_t i = 0; i < size; i++) { 7834 mEffects[i]->setDevice(device); 7835 } 7836} 7837 7838// setMode_l() must be called with PlaybackThread::mLock held 7839void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7840{ 7841 size_t size = mEffects.size(); 7842 for (size_t i = 0; i < size; i++) { 7843 mEffects[i]->setMode(mode); 7844 } 7845} 7846 7847// setVolume_l() must be called with PlaybackThread::mLock held 7848bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7849{ 7850 uint32_t newLeft = *left; 7851 uint32_t newRight = *right; 7852 bool hasControl = false; 7853 int ctrlIdx = -1; 7854 size_t size = mEffects.size(); 7855 7856 // first update volume controller 7857 for (size_t i = size; i > 0; i--) { 7858 if (mEffects[i - 1]->isProcessEnabled() && 7859 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7860 ctrlIdx = i - 1; 7861 hasControl = true; 7862 break; 7863 } 7864 } 7865 7866 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7867 if (hasControl) { 7868 *left = mNewLeftVolume; 7869 *right = mNewRightVolume; 7870 } 7871 return hasControl; 7872 } 7873 7874 mVolumeCtrlIdx = ctrlIdx; 7875 mLeftVolume = newLeft; 7876 mRightVolume = newRight; 7877 7878 // second get volume update from volume controller 7879 if (ctrlIdx >= 0) { 7880 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7881 mNewLeftVolume = newLeft; 7882 mNewRightVolume = newRight; 7883 } 7884 // then indicate volume to all other effects in chain. 7885 // Pass altered volume to effects before volume controller 7886 // and requested volume to effects after controller 7887 uint32_t lVol = newLeft; 7888 uint32_t rVol = newRight; 7889 7890 for (size_t i = 0; i < size; i++) { 7891 if ((int)i == ctrlIdx) continue; 7892 // this also works for ctrlIdx == -1 when there is no volume controller 7893 if ((int)i > ctrlIdx) { 7894 lVol = *left; 7895 rVol = *right; 7896 } 7897 mEffects[i]->setVolume(&lVol, &rVol, false); 7898 } 7899 *left = newLeft; 7900 *right = newRight; 7901 7902 return hasControl; 7903} 7904 7905status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7906{ 7907 const size_t SIZE = 256; 7908 char buffer[SIZE]; 7909 String8 result; 7910 7911 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7912 result.append(buffer); 7913 7914 bool locked = tryLock(mLock); 7915 // failed to lock - AudioFlinger is probably deadlocked 7916 if (!locked) { 7917 result.append("\tCould not lock mutex:\n"); 7918 } 7919 7920 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7921 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7922 mEffects.size(), 7923 (uint32_t)mInBuffer, 7924 (uint32_t)mOutBuffer, 7925 mActiveTrackCnt); 7926 result.append(buffer); 7927 write(fd, result.string(), result.size()); 7928 7929 for (size_t i = 0; i < mEffects.size(); ++i) { 7930 sp<EffectModule> effect = mEffects[i]; 7931 if (effect != 0) { 7932 effect->dump(fd, args); 7933 } 7934 } 7935 7936 if (locked) { 7937 mLock.unlock(); 7938 } 7939 7940 return NO_ERROR; 7941} 7942 7943// must be called with ThreadBase::mLock held 7944void AudioFlinger::EffectChain::setEffectSuspended_l( 7945 const effect_uuid_t *type, bool suspend) 7946{ 7947 sp<SuspendedEffectDesc> desc; 7948 // use effect type UUID timelow as key as there is no real risk of identical 7949 // timeLow fields among effect type UUIDs. 7950 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7951 if (suspend) { 7952 if (index >= 0) { 7953 desc = mSuspendedEffects.valueAt(index); 7954 } else { 7955 desc = new SuspendedEffectDesc(); 7956 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7957 mSuspendedEffects.add(type->timeLow, desc); 7958 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7959 } 7960 if (desc->mRefCount++ == 0) { 7961 sp<EffectModule> effect = getEffectIfEnabled(type); 7962 if (effect != 0) { 7963 desc->mEffect = effect; 7964 effect->setSuspended(true); 7965 effect->setEnabled(false); 7966 } 7967 } 7968 } else { 7969 if (index < 0) { 7970 return; 7971 } 7972 desc = mSuspendedEffects.valueAt(index); 7973 if (desc->mRefCount <= 0) { 7974 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7975 desc->mRefCount = 1; 7976 } 7977 if (--desc->mRefCount == 0) { 7978 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7979 if (desc->mEffect != 0) { 7980 sp<EffectModule> effect = desc->mEffect.promote(); 7981 if (effect != 0) { 7982 effect->setSuspended(false); 7983 sp<EffectHandle> handle = effect->controlHandle(); 7984 if (handle != 0) { 7985 effect->setEnabled(handle->enabled()); 7986 } 7987 } 7988 desc->mEffect.clear(); 7989 } 7990 mSuspendedEffects.removeItemsAt(index); 7991 } 7992 } 7993} 7994 7995// must be called with ThreadBase::mLock held 7996void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7997{ 7998 sp<SuspendedEffectDesc> desc; 7999 8000 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8001 if (suspend) { 8002 if (index >= 0) { 8003 desc = mSuspendedEffects.valueAt(index); 8004 } else { 8005 desc = new SuspendedEffectDesc(); 8006 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8007 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8008 } 8009 if (desc->mRefCount++ == 0) { 8010 Vector< sp<EffectModule> > effects; 8011 getSuspendEligibleEffects(effects); 8012 for (size_t i = 0; i < effects.size(); i++) { 8013 setEffectSuspended_l(&effects[i]->desc().type, true); 8014 } 8015 } 8016 } else { 8017 if (index < 0) { 8018 return; 8019 } 8020 desc = mSuspendedEffects.valueAt(index); 8021 if (desc->mRefCount <= 0) { 8022 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8023 desc->mRefCount = 1; 8024 } 8025 if (--desc->mRefCount == 0) { 8026 Vector<const effect_uuid_t *> types; 8027 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8028 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8029 continue; 8030 } 8031 types.add(&mSuspendedEffects.valueAt(i)->mType); 8032 } 8033 for (size_t i = 0; i < types.size(); i++) { 8034 setEffectSuspended_l(types[i], false); 8035 } 8036 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8037 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8038 } 8039 } 8040} 8041 8042 8043// The volume effect is used for automated tests only 8044#ifndef OPENSL_ES_H_ 8045static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8046 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8047const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8048#endif //OPENSL_ES_H_ 8049 8050bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8051{ 8052 // auxiliary effects and visualizer are never suspended on output mix 8053 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8054 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8055 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8056 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8057 return false; 8058 } 8059 return true; 8060} 8061 8062void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8063{ 8064 effects.clear(); 8065 for (size_t i = 0; i < mEffects.size(); i++) { 8066 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8067 effects.add(mEffects[i]); 8068 } 8069 } 8070} 8071 8072sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8073 const effect_uuid_t *type) 8074{ 8075 sp<EffectModule> effect = getEffectFromType_l(type); 8076 return effect != 0 && effect->isEnabled() ? effect : 0; 8077} 8078 8079void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8080 bool enabled) 8081{ 8082 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8083 if (enabled) { 8084 if (index < 0) { 8085 // if the effect is not suspend check if all effects are suspended 8086 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8087 if (index < 0) { 8088 return; 8089 } 8090 if (!isEffectEligibleForSuspend(effect->desc())) { 8091 return; 8092 } 8093 setEffectSuspended_l(&effect->desc().type, enabled); 8094 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8095 if (index < 0) { 8096 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8097 return; 8098 } 8099 } 8100 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8101 effect->desc().type.timeLow); 8102 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8103 // if effect is requested to suspended but was not yet enabled, supend it now. 8104 if (desc->mEffect == 0) { 8105 desc->mEffect = effect; 8106 effect->setEnabled(false); 8107 effect->setSuspended(true); 8108 } 8109 } else { 8110 if (index < 0) { 8111 return; 8112 } 8113 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8114 effect->desc().type.timeLow); 8115 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8116 desc->mEffect.clear(); 8117 effect->setSuspended(false); 8118 } 8119} 8120 8121#undef LOG_TAG 8122#define LOG_TAG "AudioFlinger" 8123 8124// ---------------------------------------------------------------------------- 8125 8126status_t AudioFlinger::onTransact( 8127 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8128{ 8129 return BnAudioFlinger::onTransact(code, data, reply, flags); 8130} 8131 8132}; // namespace android 8133