AudioFlinger.cpp revision 5cf034d92d901169ca6e36c90475f40715827fcd
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 1936bool AudioFlinger::MixerThread::threadLoop() 1937{ 1938 Vector< sp<Track> > tracksToRemove; 1939 mixer_state mixerStatus = MIXER_IDLE; 1940 nsecs_t standbyTime = systemTime(); 1941 size_t mixBufferSize = mFrameCount * mFrameSize; 1942 // FIXME: Relaxed timing because of a certain device that can't meet latency 1943 // Should be reduced to 2x after the vendor fixes the driver issue 1944 // increase threshold again due to low power audio mode. The way this warning threshold is 1945 // calculated and its usefulness should be reconsidered anyway. 1946 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1947 nsecs_t lastWarning = 0; 1948 bool longStandbyExit = false; 1949 uint32_t activeSleepTime = activeSleepTimeUs(); 1950 uint32_t idleSleepTime = idleSleepTimeUs(); 1951 uint32_t sleepTime = idleSleepTime; 1952 uint32_t sleepTimeShift = 0; 1953 Vector< sp<EffectChain> > effectChains; 1954#ifdef DEBUG_CPU_USAGE 1955 ThreadCpuUsage cpu; 1956 const CentralTendencyStatistics& stats = cpu.statistics(); 1957#endif 1958 1959 acquireWakeLock(); 1960 1961 while (!exitPending()) 1962 { 1963#ifdef DEBUG_CPU_USAGE 1964 cpu.sampleAndEnable(); 1965 unsigned n = stats.n(); 1966 // cpu.elapsed() is expensive, so don't call it every loop 1967 if ((n & 127) == 1) { 1968 long long elapsed = cpu.elapsed(); 1969 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1970 double perLoop = elapsed / (double) n; 1971 double perLoop100 = perLoop * 0.01; 1972 double mean = stats.mean(); 1973 double stddev = stats.stddev(); 1974 double minimum = stats.minimum(); 1975 double maximum = stats.maximum(); 1976 cpu.resetStatistics(); 1977 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", 1978 elapsed * .000000001, n, perLoop * .000001, 1979 mean * .001, 1980 stddev * .001, 1981 minimum * .001, 1982 maximum * .001, 1983 mean / perLoop100, 1984 stddev / perLoop100, 1985 minimum / perLoop100, 1986 maximum / perLoop100); 1987 } 1988 } 1989#endif 1990 processConfigEvents(); 1991 1992 mixerStatus = MIXER_IDLE; 1993 { // scope for mLock 1994 1995 Mutex::Autolock _l(mLock); 1996 1997 if (checkForNewParameters_l()) { 1998 mixBufferSize = mFrameCount * mFrameSize; 1999 // FIXME: Relaxed timing because of a certain device that can't meet latency 2000 // Should be reduced to 2x after the vendor fixes the driver issue 2001 // increase threshold again due to low power audio mode. The way this warning 2002 // threshold is calculated and its usefulness should be reconsidered anyway. 2003 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2004 activeSleepTime = activeSleepTimeUs(); 2005 idleSleepTime = idleSleepTimeUs(); 2006 } 2007 2008 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2009 2010 // put audio hardware into standby after short delay 2011 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2012 mSuspended)) { 2013 if (!mStandby) { 2014 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2015 mOutput->stream->common.standby(&mOutput->stream->common); 2016 mStandby = true; 2017 mBytesWritten = 0; 2018 } 2019 2020 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2021 // we're about to wait, flush the binder command buffer 2022 IPCThreadState::self()->flushCommands(); 2023 2024 if (exitPending()) break; 2025 2026 releaseWakeLock_l(); 2027 // wait until we have something to do... 2028 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 2029 mWaitWorkCV.wait(mLock); 2030 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 2031 acquireWakeLock_l(); 2032 2033 mPrevMixerStatus = MIXER_IDLE; 2034 if (!mMasterMute) { 2035 char value[PROPERTY_VALUE_MAX]; 2036 property_get("ro.audio.silent", value, "0"); 2037 if (atoi(value)) { 2038 ALOGD("Silence is golden"); 2039 setMasterMute_l(true); 2040 } 2041 } 2042 2043 standbyTime = systemTime() + mStandbyTimeInNsecs; 2044 sleepTime = idleSleepTime; 2045 sleepTimeShift = 0; 2046 continue; 2047 } 2048 } 2049 2050 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2051 2052 // prevent any changes in effect chain list and in each effect chain 2053 // during mixing and effect process as the audio buffers could be deleted 2054 // or modified if an effect is created or deleted 2055 lockEffectChains_l(effectChains); 2056 } 2057 2058 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2059 // obtain the presentation timestamp of the next output buffer 2060 int64_t pts; 2061 status_t status = INVALID_OPERATION; 2062 2063 if (NULL != mOutput->stream->get_next_write_timestamp) { 2064 status = mOutput->stream->get_next_write_timestamp( 2065 mOutput->stream, &pts); 2066 } 2067 2068 if (status != NO_ERROR) { 2069 pts = AudioBufferProvider::kInvalidPTS; 2070 } 2071 2072 // mix buffers... 2073 mAudioMixer->process(pts); 2074 // increase sleep time progressively when application underrun condition clears. 2075 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2076 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2077 // such that we would underrun the audio HAL. 2078 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2079 sleepTimeShift--; 2080 } 2081 sleepTime = 0; 2082 standbyTime = systemTime() + mStandbyTimeInNsecs; 2083 //TODO: delay standby when effects have a tail 2084 } else { 2085 // If no tracks are ready, sleep once for the duration of an output 2086 // buffer size, then write 0s to the output 2087 if (sleepTime == 0) { 2088 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2089 sleepTime = activeSleepTime >> sleepTimeShift; 2090 if (sleepTime < kMinThreadSleepTimeUs) { 2091 sleepTime = kMinThreadSleepTimeUs; 2092 } 2093 // reduce sleep time in case of consecutive application underruns to avoid 2094 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2095 // duration we would end up writing less data than needed by the audio HAL if 2096 // the condition persists. 2097 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2098 sleepTimeShift++; 2099 } 2100 } else { 2101 sleepTime = idleSleepTime; 2102 } 2103 } else if (mBytesWritten != 0 || 2104 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2105 memset (mMixBuffer, 0, mixBufferSize); 2106 sleepTime = 0; 2107 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2108 } 2109 // TODO add standby time extension fct of effect tail 2110 } 2111 2112 if (mSuspended) { 2113 sleepTime = suspendSleepTimeUs(); 2114 } 2115 // sleepTime == 0 means we must write to audio hardware 2116 if (sleepTime == 0) { 2117 for (size_t i = 0; i < effectChains.size(); i ++) { 2118 effectChains[i]->process_l(); 2119 } 2120 // enable changes in effect chain 2121 unlockEffectChains(effectChains); 2122 mLastWriteTime = systemTime(); 2123 mInWrite = true; 2124 mBytesWritten += mixBufferSize; 2125 2126 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2127 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2128 mNumWrites++; 2129 mInWrite = false; 2130 nsecs_t now = systemTime(); 2131 nsecs_t delta = now - mLastWriteTime; 2132 if (!mStandby && delta > maxPeriod) { 2133 mNumDelayedWrites++; 2134 if ((now - lastWarning) > kWarningThrottleNs) { 2135 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2136 ns2ms(delta), mNumDelayedWrites, this); 2137 lastWarning = now; 2138 } 2139 if (mStandby) { 2140 longStandbyExit = true; 2141 } 2142 } 2143 mStandby = false; 2144 } else { 2145 // enable changes in effect chain 2146 unlockEffectChains(effectChains); 2147 usleep(sleepTime); 2148 } 2149 2150 // finally let go of all our tracks, without the lock held 2151 // since we can't guarantee the destructors won't acquire that 2152 // same lock. 2153 tracksToRemove.clear(); 2154 2155 // Effect chains will be actually deleted here if they were removed from 2156 // mEffectChains list during mixing or effects processing 2157 effectChains.clear(); 2158 } 2159 2160 if (!mStandby) { 2161 mOutput->stream->common.standby(&mOutput->stream->common); 2162 } 2163 2164 releaseWakeLock(); 2165 2166 ALOGV("MixerThread %p exiting", this); 2167 return false; 2168} 2169 2170// prepareTracks_l() must be called with ThreadBase::mLock held 2171AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2172 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2173{ 2174 2175 mixer_state mixerStatus = MIXER_IDLE; 2176 // find out which tracks need to be processed 2177 size_t count = activeTracks.size(); 2178 size_t mixedTracks = 0; 2179 size_t tracksWithEffect = 0; 2180 2181 float masterVolume = mMasterVolume; 2182 bool masterMute = mMasterMute; 2183 2184 if (masterMute) { 2185 masterVolume = 0; 2186 } 2187 // Delegate master volume control to effect in output mix effect chain if needed 2188 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2189 if (chain != 0) { 2190 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2191 chain->setVolume_l(&v, &v); 2192 masterVolume = (float)((v + (1 << 23)) >> 24); 2193 chain.clear(); 2194 } 2195 2196 for (size_t i=0 ; i<count ; i++) { 2197 sp<Track> t = activeTracks[i].promote(); 2198 if (t == 0) continue; 2199 2200 // this const just means the local variable doesn't change 2201 Track* const track = t.get(); 2202 audio_track_cblk_t* cblk = track->cblk(); 2203 2204 // The first time a track is added we wait 2205 // for all its buffers to be filled before processing it 2206 int name = track->name(); 2207 // make sure that we have enough frames to mix one full buffer. 2208 // enforce this condition only once to enable draining the buffer in case the client 2209 // app does not call stop() and relies on underrun to stop: 2210 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2211 // during last round 2212 uint32_t minFrames = 1; 2213 if (!track->isStopped() && !track->isPausing() && 2214 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2215 if (t->sampleRate() == (int)mSampleRate) { 2216 minFrames = mFrameCount; 2217 } else { 2218 // +1 for rounding and +1 for additional sample needed for interpolation 2219 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2220 // add frames already consumed but not yet released by the resampler 2221 // because cblk->framesReady() will include these frames 2222 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2223 // the minimum track buffer size is normally twice the number of frames necessary 2224 // to fill one buffer and the resampler should not leave more than one buffer worth 2225 // of unreleased frames after each pass, but just in case... 2226 ALOG_ASSERT(minFrames <= cblk->frameCount); 2227 } 2228 } 2229 if ((track->framesReady() >= minFrames) && track->isReady() && 2230 !track->isPaused() && !track->isTerminated()) 2231 { 2232 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2233 2234 mixedTracks++; 2235 2236 // track->mainBuffer() != mMixBuffer means there is an effect chain 2237 // connected to the track 2238 chain.clear(); 2239 if (track->mainBuffer() != mMixBuffer) { 2240 chain = getEffectChain_l(track->sessionId()); 2241 // Delegate volume control to effect in track effect chain if needed 2242 if (chain != 0) { 2243 tracksWithEffect++; 2244 } else { 2245 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2246 name, track->sessionId()); 2247 } 2248 } 2249 2250 2251 int param = AudioMixer::VOLUME; 2252 if (track->mFillingUpStatus == Track::FS_FILLED) { 2253 // no ramp for the first volume setting 2254 track->mFillingUpStatus = Track::FS_ACTIVE; 2255 if (track->mState == TrackBase::RESUMING) { 2256 track->mState = TrackBase::ACTIVE; 2257 param = AudioMixer::RAMP_VOLUME; 2258 } 2259 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2260 } else if (cblk->server != 0) { 2261 // If the track is stopped before the first frame was mixed, 2262 // do not apply ramp 2263 param = AudioMixer::RAMP_VOLUME; 2264 } 2265 2266 // compute volume for this track 2267 uint32_t vl, vr, va; 2268 if (track->isMuted() || track->isPausing() || 2269 mStreamTypes[track->streamType()].mute) { 2270 vl = vr = va = 0; 2271 if (track->isPausing()) { 2272 track->setPaused(); 2273 } 2274 } else { 2275 2276 // read original volumes with volume control 2277 float typeVolume = mStreamTypes[track->streamType()].volume; 2278 float v = masterVolume * typeVolume; 2279 uint32_t vlr = cblk->getVolumeLR(); 2280 vl = vlr & 0xFFFF; 2281 vr = vlr >> 16; 2282 // track volumes come from shared memory, so can't be trusted and must be clamped 2283 if (vl > MAX_GAIN_INT) { 2284 ALOGV("Track left volume out of range: %04X", vl); 2285 vl = MAX_GAIN_INT; 2286 } 2287 if (vr > MAX_GAIN_INT) { 2288 ALOGV("Track right volume out of range: %04X", vr); 2289 vr = MAX_GAIN_INT; 2290 } 2291 // now apply the master volume and stream type volume 2292 vl = (uint32_t)(v * vl) << 12; 2293 vr = (uint32_t)(v * vr) << 12; 2294 // assuming master volume and stream type volume each go up to 1.0, 2295 // vl and vr are now in 8.24 format 2296 2297 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2298 // send level comes from shared memory and so may be corrupt 2299 if (sendLevel > MAX_GAIN_INT) { 2300 ALOGV("Track send level out of range: %04X", sendLevel); 2301 sendLevel = MAX_GAIN_INT; 2302 } 2303 va = (uint32_t)(v * sendLevel); 2304 } 2305 // Delegate volume control to effect in track effect chain if needed 2306 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2307 // Do not ramp volume if volume is controlled by effect 2308 param = AudioMixer::VOLUME; 2309 track->mHasVolumeController = true; 2310 } else { 2311 // force no volume ramp when volume controller was just disabled or removed 2312 // from effect chain to avoid volume spike 2313 if (track->mHasVolumeController) { 2314 param = AudioMixer::VOLUME; 2315 } 2316 track->mHasVolumeController = false; 2317 } 2318 2319 // Convert volumes from 8.24 to 4.12 format 2320 // This additional clamping is needed in case chain->setVolume_l() overshot 2321 vl = (vl + (1 << 11)) >> 12; 2322 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2323 vr = (vr + (1 << 11)) >> 12; 2324 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2325 2326 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2327 2328 // XXX: these things DON'T need to be done each time 2329 mAudioMixer->setBufferProvider(name, track); 2330 mAudioMixer->enable(name); 2331 2332 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2333 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2334 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2335 mAudioMixer->setParameter( 2336 name, 2337 AudioMixer::TRACK, 2338 AudioMixer::FORMAT, (void *)track->format()); 2339 mAudioMixer->setParameter( 2340 name, 2341 AudioMixer::TRACK, 2342 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2343 mAudioMixer->setParameter( 2344 name, 2345 AudioMixer::RESAMPLE, 2346 AudioMixer::SAMPLE_RATE, 2347 (void *)(cblk->sampleRate)); 2348 mAudioMixer->setParameter( 2349 name, 2350 AudioMixer::TRACK, 2351 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2352 mAudioMixer->setParameter( 2353 name, 2354 AudioMixer::TRACK, 2355 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2356 2357 // reset retry count 2358 track->mRetryCount = kMaxTrackRetries; 2359 // If one track is ready, set the mixer ready if: 2360 // - the mixer was not ready during previous round OR 2361 // - no other track is not ready 2362 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2363 mixerStatus != MIXER_TRACKS_ENABLED) { 2364 mixerStatus = MIXER_TRACKS_READY; 2365 } 2366 } else { 2367 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2368 if (track->isStopped()) { 2369 track->reset(); 2370 } 2371 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2372 // We have consumed all the buffers of this track. 2373 // Remove it from the list of active tracks. 2374 tracksToRemove->add(track); 2375 } else { 2376 // No buffers for this track. Give it a few chances to 2377 // fill a buffer, then remove it from active list. 2378 if (--(track->mRetryCount) <= 0) { 2379 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2380 tracksToRemove->add(track); 2381 // indicate to client process that the track was disabled because of underrun 2382 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2383 // If one track is not ready, mark the mixer also not ready if: 2384 // - the mixer was ready during previous round OR 2385 // - no other track is ready 2386 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2387 mixerStatus != MIXER_TRACKS_READY) { 2388 mixerStatus = MIXER_TRACKS_ENABLED; 2389 } 2390 } 2391 mAudioMixer->disable(name); 2392 } 2393 } 2394 2395 // remove all the tracks that need to be... 2396 count = tracksToRemove->size(); 2397 if (CC_UNLIKELY(count)) { 2398 for (size_t i=0 ; i<count ; i++) { 2399 const sp<Track>& track = tracksToRemove->itemAt(i); 2400 mActiveTracks.remove(track); 2401 if (track->mainBuffer() != mMixBuffer) { 2402 chain = getEffectChain_l(track->sessionId()); 2403 if (chain != 0) { 2404 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2405 chain->decActiveTrackCnt(); 2406 } 2407 } 2408 if (track->isTerminated()) { 2409 removeTrack_l(track); 2410 } 2411 } 2412 } 2413 2414 // mix buffer must be cleared if all tracks are connected to an 2415 // effect chain as in this case the mixer will not write to 2416 // mix buffer and track effects will accumulate into it 2417 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2418 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2419 } 2420 2421 mPrevMixerStatus = mixerStatus; 2422 return mixerStatus; 2423} 2424 2425void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2426{ 2427 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2428 this, streamType, mTracks.size()); 2429 Mutex::Autolock _l(mLock); 2430 2431 size_t size = mTracks.size(); 2432 for (size_t i = 0; i < size; i++) { 2433 sp<Track> t = mTracks[i]; 2434 if (t->streamType() == streamType) { 2435 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2436 t->mCblk->cv.signal(); 2437 } 2438 } 2439} 2440 2441void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2442{ 2443 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2444 this, streamType, valid); 2445 Mutex::Autolock _l(mLock); 2446 2447 mStreamTypes[streamType].valid = valid; 2448} 2449 2450// getTrackName_l() must be called with ThreadBase::mLock held 2451int AudioFlinger::MixerThread::getTrackName_l() 2452{ 2453 return mAudioMixer->getTrackName(); 2454} 2455 2456// deleteTrackName_l() must be called with ThreadBase::mLock held 2457void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2458{ 2459 ALOGV("remove track (%d) and delete from mixer", name); 2460 mAudioMixer->deleteTrackName(name); 2461} 2462 2463// checkForNewParameters_l() must be called with ThreadBase::mLock held 2464bool AudioFlinger::MixerThread::checkForNewParameters_l() 2465{ 2466 bool reconfig = false; 2467 2468 while (!mNewParameters.isEmpty()) { 2469 status_t status = NO_ERROR; 2470 String8 keyValuePair = mNewParameters[0]; 2471 AudioParameter param = AudioParameter(keyValuePair); 2472 int value; 2473 2474 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2475 reconfig = true; 2476 } 2477 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2478 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2479 status = BAD_VALUE; 2480 } else { 2481 reconfig = true; 2482 } 2483 } 2484 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2485 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2486 status = BAD_VALUE; 2487 } else { 2488 reconfig = true; 2489 } 2490 } 2491 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2492 // do not accept frame count changes if tracks are open as the track buffer 2493 // size depends on frame count and correct behavior would not be guaranteed 2494 // if frame count is changed after track creation 2495 if (!mTracks.isEmpty()) { 2496 status = INVALID_OPERATION; 2497 } else { 2498 reconfig = true; 2499 } 2500 } 2501 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2502 // when changing the audio output device, call addBatteryData to notify 2503 // the change 2504 if ((int)mDevice != value) { 2505 uint32_t params = 0; 2506 // check whether speaker is on 2507 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2508 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2509 } 2510 2511 int deviceWithoutSpeaker 2512 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2513 // check if any other device (except speaker) is on 2514 if (value & deviceWithoutSpeaker ) { 2515 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2516 } 2517 2518 if (params != 0) { 2519 addBatteryData(params); 2520 } 2521 } 2522 2523 // forward device change to effects that have requested to be 2524 // aware of attached audio device. 2525 mDevice = (uint32_t)value; 2526 for (size_t i = 0; i < mEffectChains.size(); i++) { 2527 mEffectChains[i]->setDevice_l(mDevice); 2528 } 2529 } 2530 2531 if (status == NO_ERROR) { 2532 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2533 keyValuePair.string()); 2534 if (!mStandby && status == INVALID_OPERATION) { 2535 mOutput->stream->common.standby(&mOutput->stream->common); 2536 mStandby = true; 2537 mBytesWritten = 0; 2538 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2539 keyValuePair.string()); 2540 } 2541 if (status == NO_ERROR && reconfig) { 2542 delete mAudioMixer; 2543 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2544 mAudioMixer = NULL; 2545 readOutputParameters(); 2546 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2547 for (size_t i = 0; i < mTracks.size() ; i++) { 2548 int name = getTrackName_l(); 2549 if (name < 0) break; 2550 mTracks[i]->mName = name; 2551 // limit track sample rate to 2 x new output sample rate 2552 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2553 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2554 } 2555 } 2556 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2557 } 2558 } 2559 2560 mNewParameters.removeAt(0); 2561 2562 mParamStatus = status; 2563 mParamCond.signal(); 2564 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2565 // already timed out waiting for the status and will never signal the condition. 2566 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2567 } 2568 return reconfig; 2569} 2570 2571status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2572{ 2573 const size_t SIZE = 256; 2574 char buffer[SIZE]; 2575 String8 result; 2576 2577 PlaybackThread::dumpInternals(fd, args); 2578 2579 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2580 result.append(buffer); 2581 write(fd, result.string(), result.size()); 2582 return NO_ERROR; 2583} 2584 2585uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2586{ 2587 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2588} 2589 2590uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2591{ 2592 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2593} 2594 2595// ---------------------------------------------------------------------------- 2596AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2597 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2598 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2599 // mLeftVolFloat, mRightVolFloat 2600 // mLeftVolShort, mRightVolShort 2601{ 2602} 2603 2604AudioFlinger::DirectOutputThread::~DirectOutputThread() 2605{ 2606} 2607 2608void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2609{ 2610 // Do not apply volume on compressed audio 2611 if (!audio_is_linear_pcm(mFormat)) { 2612 return; 2613 } 2614 2615 // convert to signed 16 bit before volume calculation 2616 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2617 size_t count = mFrameCount * mChannelCount; 2618 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2619 int16_t *dst = mMixBuffer + count-1; 2620 while(count--) { 2621 *dst-- = (int16_t)(*src--^0x80) << 8; 2622 } 2623 } 2624 2625 size_t frameCount = mFrameCount; 2626 int16_t *out = mMixBuffer; 2627 if (ramp) { 2628 if (mChannelCount == 1) { 2629 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2630 int32_t vlInc = d / (int32_t)frameCount; 2631 int32_t vl = ((int32_t)mLeftVolShort << 16); 2632 do { 2633 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2634 out++; 2635 vl += vlInc; 2636 } while (--frameCount); 2637 2638 } else { 2639 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2640 int32_t vlInc = d / (int32_t)frameCount; 2641 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2642 int32_t vrInc = d / (int32_t)frameCount; 2643 int32_t vl = ((int32_t)mLeftVolShort << 16); 2644 int32_t vr = ((int32_t)mRightVolShort << 16); 2645 do { 2646 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2647 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2648 out += 2; 2649 vl += vlInc; 2650 vr += vrInc; 2651 } while (--frameCount); 2652 } 2653 } else { 2654 if (mChannelCount == 1) { 2655 do { 2656 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2657 out++; 2658 } while (--frameCount); 2659 } else { 2660 do { 2661 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2662 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2663 out += 2; 2664 } while (--frameCount); 2665 } 2666 } 2667 2668 // convert back to unsigned 8 bit after volume calculation 2669 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2670 size_t count = mFrameCount * mChannelCount; 2671 int16_t *src = mMixBuffer; 2672 uint8_t *dst = (uint8_t *)mMixBuffer; 2673 while(count--) { 2674 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2675 } 2676 } 2677 2678 mLeftVolShort = leftVol; 2679 mRightVolShort = rightVol; 2680} 2681 2682bool AudioFlinger::DirectOutputThread::threadLoop() 2683{ 2684 mixer_state mixerStatus = MIXER_IDLE; 2685 sp<Track> trackToRemove; 2686 sp<Track> activeTrack; 2687 nsecs_t standbyTime = systemTime(); 2688 size_t mixBufferSize = mFrameCount*mFrameSize; 2689 uint32_t activeSleepTime = activeSleepTimeUs(); 2690 uint32_t idleSleepTime = idleSleepTimeUs(); 2691 uint32_t sleepTime = idleSleepTime; 2692 // use shorter standby delay as on normal output to release 2693 // hardware resources as soon as possible 2694 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2695 2696 acquireWakeLock(); 2697 2698 while (!exitPending()) 2699 { 2700 bool rampVolume; 2701 uint16_t leftVol; 2702 uint16_t rightVol; 2703 Vector< sp<EffectChain> > effectChains; 2704 2705 processConfigEvents(); 2706 2707 mixerStatus = MIXER_IDLE; 2708 2709 { // scope for the mLock 2710 2711 Mutex::Autolock _l(mLock); 2712 2713 if (checkForNewParameters_l()) { 2714 mixBufferSize = mFrameCount*mFrameSize; 2715 activeSleepTime = activeSleepTimeUs(); 2716 idleSleepTime = idleSleepTimeUs(); 2717 standbyDelay = microseconds(activeSleepTime*2); 2718 } 2719 2720 // put audio hardware into standby after short delay 2721 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2722 mSuspended)) { 2723 // wait until we have something to do... 2724 if (!mStandby) { 2725 ALOGV("Audio hardware entering standby, mixer %p", this); 2726 mOutput->stream->common.standby(&mOutput->stream->common); 2727 mStandby = true; 2728 mBytesWritten = 0; 2729 } 2730 2731 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2732 // we're about to wait, flush the binder command buffer 2733 IPCThreadState::self()->flushCommands(); 2734 2735 if (exitPending()) break; 2736 2737 releaseWakeLock_l(); 2738 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2739 mWaitWorkCV.wait(mLock); 2740 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2741 acquireWakeLock_l(); 2742 2743 if (!mMasterMute) { 2744 char value[PROPERTY_VALUE_MAX]; 2745 property_get("ro.audio.silent", value, "0"); 2746 if (atoi(value)) { 2747 ALOGD("Silence is golden"); 2748 setMasterMute_l(true); 2749 } 2750 } 2751 2752 standbyTime = systemTime() + standbyDelay; 2753 sleepTime = idleSleepTime; 2754 continue; 2755 } 2756 } 2757 2758 effectChains = mEffectChains; 2759 2760 // find out which tracks need to be processed 2761 if (mActiveTracks.size() != 0) { 2762 sp<Track> t = mActiveTracks[0].promote(); 2763 if (t == 0) continue; 2764 2765 Track* const track = t.get(); 2766 audio_track_cblk_t* cblk = track->cblk(); 2767 2768 // The first time a track is added we wait 2769 // for all its buffers to be filled before processing it 2770 if (cblk->framesReady() && track->isReady() && 2771 !track->isPaused() && !track->isTerminated()) 2772 { 2773 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2774 2775 if (track->mFillingUpStatus == Track::FS_FILLED) { 2776 track->mFillingUpStatus = Track::FS_ACTIVE; 2777 mLeftVolFloat = mRightVolFloat = 0; 2778 mLeftVolShort = mRightVolShort = 0; 2779 if (track->mState == TrackBase::RESUMING) { 2780 track->mState = TrackBase::ACTIVE; 2781 rampVolume = true; 2782 } 2783 } else if (cblk->server != 0) { 2784 // If the track is stopped before the first frame was mixed, 2785 // do not apply ramp 2786 rampVolume = true; 2787 } 2788 // compute volume for this track 2789 float left, right; 2790 if (track->isMuted() || mMasterMute || track->isPausing() || 2791 mStreamTypes[track->streamType()].mute) { 2792 left = right = 0; 2793 if (track->isPausing()) { 2794 track->setPaused(); 2795 } 2796 } else { 2797 float typeVolume = mStreamTypes[track->streamType()].volume; 2798 float v = mMasterVolume * typeVolume; 2799 uint32_t vlr = cblk->getVolumeLR(); 2800 float v_clamped = v * (vlr & 0xFFFF); 2801 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2802 left = v_clamped/MAX_GAIN; 2803 v_clamped = v * (vlr >> 16); 2804 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2805 right = v_clamped/MAX_GAIN; 2806 } 2807 2808 if (left != mLeftVolFloat || right != mRightVolFloat) { 2809 mLeftVolFloat = left; 2810 mRightVolFloat = right; 2811 2812 // If audio HAL implements volume control, 2813 // force software volume to nominal value 2814 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2815 left = 1.0f; 2816 right = 1.0f; 2817 } 2818 2819 // Convert volumes from float to 8.24 2820 uint32_t vl = (uint32_t)(left * (1 << 24)); 2821 uint32_t vr = (uint32_t)(right * (1 << 24)); 2822 2823 // Delegate volume control to effect in track effect chain if needed 2824 // only one effect chain can be present on DirectOutputThread, so if 2825 // there is one, the track is connected to it 2826 if (!effectChains.isEmpty()) { 2827 // Do not ramp volume if volume is controlled by effect 2828 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2829 rampVolume = false; 2830 } 2831 } 2832 2833 // Convert volumes from 8.24 to 4.12 format 2834 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2835 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2836 leftVol = (uint16_t)v_clamped; 2837 v_clamped = (vr + (1 << 11)) >> 12; 2838 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2839 rightVol = (uint16_t)v_clamped; 2840 } else { 2841 leftVol = mLeftVolShort; 2842 rightVol = mRightVolShort; 2843 rampVolume = false; 2844 } 2845 2846 // reset retry count 2847 track->mRetryCount = kMaxTrackRetriesDirect; 2848 activeTrack = t; 2849 mixerStatus = MIXER_TRACKS_READY; 2850 } else { 2851 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2852 if (track->isStopped()) { 2853 track->reset(); 2854 } 2855 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2856 // We have consumed all the buffers of this track. 2857 // Remove it from the list of active tracks. 2858 trackToRemove = track; 2859 } else { 2860 // No buffers for this track. Give it a few chances to 2861 // fill a buffer, then remove it from active list. 2862 if (--(track->mRetryCount) <= 0) { 2863 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2864 trackToRemove = track; 2865 } else { 2866 mixerStatus = MIXER_TRACKS_ENABLED; 2867 } 2868 } 2869 } 2870 } 2871 2872 // remove all the tracks that need to be... 2873 if (CC_UNLIKELY(trackToRemove != 0)) { 2874 mActiveTracks.remove(trackToRemove); 2875 if (!effectChains.isEmpty()) { 2876 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2877 trackToRemove->sessionId()); 2878 effectChains[0]->decActiveTrackCnt(); 2879 } 2880 if (trackToRemove->isTerminated()) { 2881 removeTrack_l(trackToRemove); 2882 } 2883 } 2884 2885 lockEffectChains_l(effectChains); 2886 } 2887 2888 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2889 AudioBufferProvider::Buffer buffer; 2890 size_t frameCount = mFrameCount; 2891 int8_t *curBuf = (int8_t *)mMixBuffer; 2892 // output audio to hardware 2893 while (frameCount) { 2894 buffer.frameCount = frameCount; 2895 activeTrack->getNextBuffer(&buffer, 2896 AudioBufferProvider::kInvalidPTS); 2897 if (CC_UNLIKELY(buffer.raw == NULL)) { 2898 memset(curBuf, 0, frameCount * mFrameSize); 2899 break; 2900 } 2901 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2902 frameCount -= buffer.frameCount; 2903 curBuf += buffer.frameCount * mFrameSize; 2904 activeTrack->releaseBuffer(&buffer); 2905 } 2906 sleepTime = 0; 2907 standbyTime = systemTime() + standbyDelay; 2908 } else { 2909 if (sleepTime == 0) { 2910 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2911 sleepTime = activeSleepTime; 2912 } else { 2913 sleepTime = idleSleepTime; 2914 } 2915 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2916 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2917 sleepTime = 0; 2918 } 2919 } 2920 2921 if (mSuspended) { 2922 sleepTime = suspendSleepTimeUs(); 2923 } 2924 // sleepTime == 0 means we must write to audio hardware 2925 if (sleepTime == 0) { 2926 if (mixerStatus == MIXER_TRACKS_READY) { 2927 applyVolume(leftVol, rightVol, rampVolume); 2928 } 2929 for (size_t i = 0; i < effectChains.size(); i ++) { 2930 effectChains[i]->process_l(); 2931 } 2932 unlockEffectChains(effectChains); 2933 2934 mLastWriteTime = systemTime(); 2935 mInWrite = true; 2936 mBytesWritten += mixBufferSize; 2937 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2938 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2939 mNumWrites++; 2940 mInWrite = false; 2941 mStandby = false; 2942 } else { 2943 unlockEffectChains(effectChains); 2944 usleep(sleepTime); 2945 } 2946 2947 // finally let go of removed track, without the lock held 2948 // since we can't guarantee the destructors won't acquire that 2949 // same lock. 2950 trackToRemove.clear(); 2951 activeTrack.clear(); 2952 2953 // Effect chains will be actually deleted here if they were removed from 2954 // mEffectChains list during mixing or effects processing 2955 effectChains.clear(); 2956 } 2957 2958 if (!mStandby) { 2959 mOutput->stream->common.standby(&mOutput->stream->common); 2960 } 2961 2962 releaseWakeLock(); 2963 2964 ALOGV("DirectOutputThread %p exiting", this); 2965 return false; 2966} 2967 2968// getTrackName_l() must be called with ThreadBase::mLock held 2969int AudioFlinger::DirectOutputThread::getTrackName_l() 2970{ 2971 return 0; 2972} 2973 2974// deleteTrackName_l() must be called with ThreadBase::mLock held 2975void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2976{ 2977} 2978 2979// checkForNewParameters_l() must be called with ThreadBase::mLock held 2980bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2981{ 2982 bool reconfig = false; 2983 2984 while (!mNewParameters.isEmpty()) { 2985 status_t status = NO_ERROR; 2986 String8 keyValuePair = mNewParameters[0]; 2987 AudioParameter param = AudioParameter(keyValuePair); 2988 int value; 2989 2990 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2991 // do not accept frame count changes if tracks are open as the track buffer 2992 // size depends on frame count and correct behavior would not be garantied 2993 // if frame count is changed after track creation 2994 if (!mTracks.isEmpty()) { 2995 status = INVALID_OPERATION; 2996 } else { 2997 reconfig = true; 2998 } 2999 } 3000 if (status == NO_ERROR) { 3001 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3002 keyValuePair.string()); 3003 if (!mStandby && status == INVALID_OPERATION) { 3004 mOutput->stream->common.standby(&mOutput->stream->common); 3005 mStandby = true; 3006 mBytesWritten = 0; 3007 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3008 keyValuePair.string()); 3009 } 3010 if (status == NO_ERROR && reconfig) { 3011 readOutputParameters(); 3012 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3013 } 3014 } 3015 3016 mNewParameters.removeAt(0); 3017 3018 mParamStatus = status; 3019 mParamCond.signal(); 3020 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3021 // already timed out waiting for the status and will never signal the condition. 3022 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3023 } 3024 return reconfig; 3025} 3026 3027uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3028{ 3029 uint32_t time; 3030 if (audio_is_linear_pcm(mFormat)) { 3031 time = PlaybackThread::activeSleepTimeUs(); 3032 } else { 3033 time = 10000; 3034 } 3035 return time; 3036} 3037 3038uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3039{ 3040 uint32_t time; 3041 if (audio_is_linear_pcm(mFormat)) { 3042 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3043 } else { 3044 time = 10000; 3045 } 3046 return time; 3047} 3048 3049uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3050{ 3051 uint32_t time; 3052 if (audio_is_linear_pcm(mFormat)) { 3053 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3054 } else { 3055 time = 10000; 3056 } 3057 return time; 3058} 3059 3060 3061// ---------------------------------------------------------------------------- 3062 3063AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3064 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3065 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3066 mWaitTimeMs(UINT_MAX) 3067{ 3068 addOutputTrack(mainThread); 3069} 3070 3071AudioFlinger::DuplicatingThread::~DuplicatingThread() 3072{ 3073 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3074 mOutputTracks[i]->destroy(); 3075 } 3076} 3077 3078bool AudioFlinger::DuplicatingThread::threadLoop() 3079{ 3080 Vector< sp<Track> > tracksToRemove; 3081 mixer_state mixerStatus = MIXER_IDLE; 3082 nsecs_t standbyTime = systemTime(); 3083 size_t mixBufferSize = mFrameCount*mFrameSize; 3084 SortedVector< sp<OutputTrack> > outputTracks; 3085 uint32_t writeFrames = 0; 3086 uint32_t activeSleepTime = activeSleepTimeUs(); 3087 uint32_t idleSleepTime = idleSleepTimeUs(); 3088 uint32_t sleepTime = idleSleepTime; 3089 Vector< sp<EffectChain> > effectChains; 3090 3091 acquireWakeLock(); 3092 3093 while (!exitPending()) 3094 { 3095 processConfigEvents(); 3096 3097 mixerStatus = MIXER_IDLE; 3098 { // scope for the mLock 3099 3100 Mutex::Autolock _l(mLock); 3101 3102 if (checkForNewParameters_l()) { 3103 mixBufferSize = mFrameCount*mFrameSize; 3104 updateWaitTime(); 3105 activeSleepTime = activeSleepTimeUs(); 3106 idleSleepTime = idleSleepTimeUs(); 3107 } 3108 3109 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3110 3111 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3112 outputTracks.add(mOutputTracks[i]); 3113 } 3114 3115 // put audio hardware into standby after short delay 3116 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3117 mSuspended)) { 3118 if (!mStandby) { 3119 for (size_t i = 0; i < outputTracks.size(); i++) { 3120 outputTracks[i]->stop(); 3121 } 3122 mStandby = true; 3123 mBytesWritten = 0; 3124 } 3125 3126 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3127 // we're about to wait, flush the binder command buffer 3128 IPCThreadState::self()->flushCommands(); 3129 outputTracks.clear(); 3130 3131 if (exitPending()) break; 3132 3133 releaseWakeLock_l(); 3134 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3135 mWaitWorkCV.wait(mLock); 3136 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3137 acquireWakeLock_l(); 3138 3139 mPrevMixerStatus = MIXER_IDLE; 3140 if (!mMasterMute) { 3141 char value[PROPERTY_VALUE_MAX]; 3142 property_get("ro.audio.silent", value, "0"); 3143 if (atoi(value)) { 3144 ALOGD("Silence is golden"); 3145 setMasterMute_l(true); 3146 } 3147 } 3148 3149 standbyTime = systemTime() + mStandbyTimeInNsecs; 3150 sleepTime = idleSleepTime; 3151 continue; 3152 } 3153 } 3154 3155 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3156 3157 // prevent any changes in effect chain list and in each effect chain 3158 // during mixing and effect process as the audio buffers could be deleted 3159 // or modified if an effect is created or deleted 3160 lockEffectChains_l(effectChains); 3161 } 3162 3163 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3164 // mix buffers... 3165 if (outputsReady(outputTracks)) { 3166 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3167 } else { 3168 memset(mMixBuffer, 0, mixBufferSize); 3169 } 3170 sleepTime = 0; 3171 writeFrames = mFrameCount; 3172 } else { 3173 if (sleepTime == 0) { 3174 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3175 sleepTime = activeSleepTime; 3176 } else { 3177 sleepTime = idleSleepTime; 3178 } 3179 } else if (mBytesWritten != 0) { 3180 // flush remaining overflow buffers in output tracks 3181 for (size_t i = 0; i < outputTracks.size(); i++) { 3182 if (outputTracks[i]->isActive()) { 3183 sleepTime = 0; 3184 writeFrames = 0; 3185 memset(mMixBuffer, 0, mixBufferSize); 3186 break; 3187 } 3188 } 3189 } 3190 } 3191 3192 if (mSuspended) { 3193 sleepTime = suspendSleepTimeUs(); 3194 } 3195 // sleepTime == 0 means we must write to audio hardware 3196 if (sleepTime == 0) { 3197 for (size_t i = 0; i < effectChains.size(); i ++) { 3198 effectChains[i]->process_l(); 3199 } 3200 // enable changes in effect chain 3201 unlockEffectChains(effectChains); 3202 3203 standbyTime = systemTime() + mStandbyTimeInNsecs; 3204 for (size_t i = 0; i < outputTracks.size(); i++) { 3205 outputTracks[i]->write(mMixBuffer, writeFrames); 3206 } 3207 mStandby = false; 3208 mBytesWritten += mixBufferSize; 3209 } else { 3210 // enable changes in effect chain 3211 unlockEffectChains(effectChains); 3212 usleep(sleepTime); 3213 } 3214 3215 // finally let go of all our tracks, without the lock held 3216 // since we can't guarantee the destructors won't acquire that 3217 // same lock. 3218 tracksToRemove.clear(); 3219 outputTracks.clear(); 3220 3221 // Effect chains will be actually deleted here if they were removed from 3222 // mEffectChains list during mixing or effects processing 3223 effectChains.clear(); 3224 } 3225 3226 releaseWakeLock(); 3227 3228 return false; 3229} 3230 3231void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3232{ 3233 // FIXME explain this formula 3234 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3235 OutputTrack *outputTrack = new OutputTrack(thread, 3236 this, 3237 mSampleRate, 3238 mFormat, 3239 mChannelMask, 3240 frameCount); 3241 if (outputTrack->cblk() != NULL) { 3242 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3243 mOutputTracks.add(outputTrack); 3244 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3245 updateWaitTime(); 3246 } 3247} 3248 3249void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3250{ 3251 Mutex::Autolock _l(mLock); 3252 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3253 if (mOutputTracks[i]->thread() == thread) { 3254 mOutputTracks[i]->destroy(); 3255 mOutputTracks.removeAt(i); 3256 updateWaitTime(); 3257 return; 3258 } 3259 } 3260 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3261} 3262 3263void AudioFlinger::DuplicatingThread::updateWaitTime() 3264{ 3265 mWaitTimeMs = UINT_MAX; 3266 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3267 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3268 if (strong != 0) { 3269 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3270 if (waitTimeMs < mWaitTimeMs) { 3271 mWaitTimeMs = waitTimeMs; 3272 } 3273 } 3274 } 3275} 3276 3277 3278bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3279{ 3280 for (size_t i = 0; i < outputTracks.size(); i++) { 3281 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3282 if (thread == 0) { 3283 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3284 return false; 3285 } 3286 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3287 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3288 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3289 return false; 3290 } 3291 } 3292 return true; 3293} 3294 3295uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3296{ 3297 return (mWaitTimeMs * 1000) / 2; 3298} 3299 3300// ---------------------------------------------------------------------------- 3301 3302// TrackBase constructor must be called with AudioFlinger::mLock held 3303AudioFlinger::ThreadBase::TrackBase::TrackBase( 3304 ThreadBase *thread, 3305 const sp<Client>& client, 3306 uint32_t sampleRate, 3307 audio_format_t format, 3308 uint32_t channelMask, 3309 int frameCount, 3310 const sp<IMemory>& sharedBuffer, 3311 int sessionId) 3312 : RefBase(), 3313 mThread(thread), 3314 mClient(client), 3315 mCblk(NULL), 3316 // mBuffer 3317 // mBufferEnd 3318 mFrameCount(0), 3319 mState(IDLE), 3320 mFormat(format), 3321 mStepServerFailed(false), 3322 mSessionId(sessionId) 3323 // mChannelCount 3324 // mChannelMask 3325{ 3326 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3327 3328 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3329 size_t size = sizeof(audio_track_cblk_t); 3330 uint8_t channelCount = popcount(channelMask); 3331 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3332 if (sharedBuffer == 0) { 3333 size += bufferSize; 3334 } 3335 3336 if (client != NULL) { 3337 mCblkMemory = client->heap()->allocate(size); 3338 if (mCblkMemory != 0) { 3339 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3340 if (mCblk != NULL) { // construct the shared structure in-place. 3341 new(mCblk) audio_track_cblk_t(); 3342 // clear all buffers 3343 mCblk->frameCount = frameCount; 3344 mCblk->sampleRate = sampleRate; 3345 mChannelCount = channelCount; 3346 mChannelMask = channelMask; 3347 if (sharedBuffer == 0) { 3348 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3349 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3350 // Force underrun condition to avoid false underrun callback until first data is 3351 // written to buffer (other flags are cleared) 3352 mCblk->flags = CBLK_UNDERRUN_ON; 3353 } else { 3354 mBuffer = sharedBuffer->pointer(); 3355 } 3356 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3357 } 3358 } else { 3359 ALOGE("not enough memory for AudioTrack size=%u", size); 3360 client->heap()->dump("AudioTrack"); 3361 return; 3362 } 3363 } else { 3364 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3365 // construct the shared structure in-place. 3366 new(mCblk) audio_track_cblk_t(); 3367 // clear all buffers 3368 mCblk->frameCount = frameCount; 3369 mCblk->sampleRate = sampleRate; 3370 mChannelCount = channelCount; 3371 mChannelMask = channelMask; 3372 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3373 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3374 // Force underrun condition to avoid false underrun callback until first data is 3375 // written to buffer (other flags are cleared) 3376 mCblk->flags = CBLK_UNDERRUN_ON; 3377 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3378 } 3379} 3380 3381AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3382{ 3383 if (mCblk != NULL) { 3384 if (mClient == 0) { 3385 delete mCblk; 3386 } else { 3387 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3388 } 3389 } 3390 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3391 if (mClient != 0) { 3392 // Client destructor must run with AudioFlinger mutex locked 3393 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3394 // If the client's reference count drops to zero, the associated destructor 3395 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3396 // relying on the automatic clear() at end of scope. 3397 mClient.clear(); 3398 } 3399} 3400 3401void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3402{ 3403 buffer->raw = NULL; 3404 mFrameCount = buffer->frameCount; 3405 step(); 3406 buffer->frameCount = 0; 3407} 3408 3409bool AudioFlinger::ThreadBase::TrackBase::step() { 3410 bool result; 3411 audio_track_cblk_t* cblk = this->cblk(); 3412 3413 result = cblk->stepServer(mFrameCount); 3414 if (!result) { 3415 ALOGV("stepServer failed acquiring cblk mutex"); 3416 mStepServerFailed = true; 3417 } 3418 return result; 3419} 3420 3421void AudioFlinger::ThreadBase::TrackBase::reset() { 3422 audio_track_cblk_t* cblk = this->cblk(); 3423 3424 cblk->user = 0; 3425 cblk->server = 0; 3426 cblk->userBase = 0; 3427 cblk->serverBase = 0; 3428 mStepServerFailed = false; 3429 ALOGV("TrackBase::reset"); 3430} 3431 3432int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3433 return (int)mCblk->sampleRate; 3434} 3435 3436void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3437 audio_track_cblk_t* cblk = this->cblk(); 3438 size_t frameSize = cblk->frameSize; 3439 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3440 int8_t *bufferEnd = bufferStart + frames * frameSize; 3441 3442 // Check validity of returned pointer in case the track control block would have been corrupted. 3443 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3444 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3445 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3446 server %d, serverBase %d, user %d, userBase %d", 3447 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3448 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3449 return NULL; 3450 } 3451 3452 return bufferStart; 3453} 3454 3455// ---------------------------------------------------------------------------- 3456 3457// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3458AudioFlinger::PlaybackThread::Track::Track( 3459 PlaybackThread *thread, 3460 const sp<Client>& client, 3461 audio_stream_type_t streamType, 3462 uint32_t sampleRate, 3463 audio_format_t format, 3464 uint32_t channelMask, 3465 int frameCount, 3466 const sp<IMemory>& sharedBuffer, 3467 int sessionId) 3468 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3469 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3470 mAuxEffectId(0), mHasVolumeController(false) 3471{ 3472 if (mCblk != NULL) { 3473 if (thread != NULL) { 3474 mName = thread->getTrackName_l(); 3475 mMainBuffer = thread->mixBuffer(); 3476 } 3477 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3478 if (mName < 0) { 3479 ALOGE("no more track names available"); 3480 } 3481 mStreamType = streamType; 3482 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3483 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3484 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3485 } 3486} 3487 3488AudioFlinger::PlaybackThread::Track::~Track() 3489{ 3490 ALOGV("PlaybackThread::Track destructor"); 3491 sp<ThreadBase> thread = mThread.promote(); 3492 if (thread != 0) { 3493 Mutex::Autolock _l(thread->mLock); 3494 mState = TERMINATED; 3495 } 3496} 3497 3498void AudioFlinger::PlaybackThread::Track::destroy() 3499{ 3500 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3501 // by removing it from mTracks vector, so there is a risk that this Tracks's 3502 // destructor is called. As the destructor needs to lock mLock, 3503 // we must acquire a strong reference on this Track before locking mLock 3504 // here so that the destructor is called only when exiting this function. 3505 // On the other hand, as long as Track::destroy() is only called by 3506 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3507 // this Track with its member mTrack. 3508 sp<Track> keep(this); 3509 { // scope for mLock 3510 sp<ThreadBase> thread = mThread.promote(); 3511 if (thread != 0) { 3512 if (!isOutputTrack()) { 3513 if (mState == ACTIVE || mState == RESUMING) { 3514 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3515 3516 // to track the speaker usage 3517 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3518 } 3519 AudioSystem::releaseOutput(thread->id()); 3520 } 3521 Mutex::Autolock _l(thread->mLock); 3522 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3523 playbackThread->destroyTrack_l(this); 3524 } 3525 } 3526} 3527 3528void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3529{ 3530 uint32_t vlr = mCblk->getVolumeLR(); 3531 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", 3532 mName - AudioMixer::TRACK0, 3533 (mClient == 0) ? getpid_cached : mClient->pid(), 3534 mStreamType, 3535 mFormat, 3536 mChannelMask, 3537 mSessionId, 3538 mFrameCount, 3539 mState, 3540 mMute, 3541 mFillingUpStatus, 3542 mCblk->sampleRate, 3543 vlr & 0xFFFF, 3544 vlr >> 16, 3545 mCblk->server, 3546 mCblk->user, 3547 (int)mMainBuffer, 3548 (int)mAuxBuffer); 3549} 3550 3551status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3552 AudioBufferProvider::Buffer* buffer, int64_t pts) 3553{ 3554 audio_track_cblk_t* cblk = this->cblk(); 3555 uint32_t framesReady; 3556 uint32_t framesReq = buffer->frameCount; 3557 3558 // Check if last stepServer failed, try to step now 3559 if (mStepServerFailed) { 3560 if (!step()) goto getNextBuffer_exit; 3561 ALOGV("stepServer recovered"); 3562 mStepServerFailed = false; 3563 } 3564 3565 framesReady = cblk->framesReady(); 3566 3567 if (CC_LIKELY(framesReady)) { 3568 uint32_t s = cblk->server; 3569 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3570 3571 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3572 if (framesReq > framesReady) { 3573 framesReq = framesReady; 3574 } 3575 if (s + framesReq > bufferEnd) { 3576 framesReq = bufferEnd - s; 3577 } 3578 3579 buffer->raw = getBuffer(s, framesReq); 3580 if (buffer->raw == NULL) goto getNextBuffer_exit; 3581 3582 buffer->frameCount = framesReq; 3583 return NO_ERROR; 3584 } 3585 3586getNextBuffer_exit: 3587 buffer->raw = NULL; 3588 buffer->frameCount = 0; 3589 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3590 return NOT_ENOUGH_DATA; 3591} 3592 3593uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3594 return mCblk->framesReady(); 3595} 3596 3597bool AudioFlinger::PlaybackThread::Track::isReady() const { 3598 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3599 3600 if (framesReady() >= mCblk->frameCount || 3601 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3602 mFillingUpStatus = FS_FILLED; 3603 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3604 return true; 3605 } 3606 return false; 3607} 3608 3609status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3610{ 3611 status_t status = NO_ERROR; 3612 ALOGV("start(%d), calling pid %d session %d tid %d", 3613 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3614 sp<ThreadBase> thread = mThread.promote(); 3615 if (thread != 0) { 3616 Mutex::Autolock _l(thread->mLock); 3617 track_state state = mState; 3618 // here the track could be either new, or restarted 3619 // in both cases "unstop" the track 3620 if (mState == PAUSED) { 3621 mState = TrackBase::RESUMING; 3622 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3623 } else { 3624 mState = TrackBase::ACTIVE; 3625 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3626 } 3627 3628 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3629 thread->mLock.unlock(); 3630 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3631 thread->mLock.lock(); 3632 3633 // to track the speaker usage 3634 if (status == NO_ERROR) { 3635 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3636 } 3637 } 3638 if (status == NO_ERROR) { 3639 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3640 playbackThread->addTrack_l(this); 3641 } else { 3642 mState = state; 3643 } 3644 } else { 3645 status = BAD_VALUE; 3646 } 3647 return status; 3648} 3649 3650void AudioFlinger::PlaybackThread::Track::stop() 3651{ 3652 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3653 sp<ThreadBase> thread = mThread.promote(); 3654 if (thread != 0) { 3655 Mutex::Autolock _l(thread->mLock); 3656 track_state state = mState; 3657 if (mState > STOPPED) { 3658 mState = STOPPED; 3659 // If the track is not active (PAUSED and buffers full), flush buffers 3660 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3661 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3662 reset(); 3663 } 3664 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3665 } 3666 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3667 thread->mLock.unlock(); 3668 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3669 thread->mLock.lock(); 3670 3671 // to track the speaker usage 3672 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3673 } 3674 } 3675} 3676 3677void AudioFlinger::PlaybackThread::Track::pause() 3678{ 3679 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3680 sp<ThreadBase> thread = mThread.promote(); 3681 if (thread != 0) { 3682 Mutex::Autolock _l(thread->mLock); 3683 if (mState == ACTIVE || mState == RESUMING) { 3684 mState = PAUSING; 3685 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3686 if (!isOutputTrack()) { 3687 thread->mLock.unlock(); 3688 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3689 thread->mLock.lock(); 3690 3691 // to track the speaker usage 3692 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3693 } 3694 } 3695 } 3696} 3697 3698void AudioFlinger::PlaybackThread::Track::flush() 3699{ 3700 ALOGV("flush(%d)", mName); 3701 sp<ThreadBase> thread = mThread.promote(); 3702 if (thread != 0) { 3703 Mutex::Autolock _l(thread->mLock); 3704 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3705 return; 3706 } 3707 // No point remaining in PAUSED state after a flush => go to 3708 // STOPPED state 3709 mState = STOPPED; 3710 3711 // do not reset the track if it is still in the process of being stopped or paused. 3712 // this will be done by prepareTracks_l() when the track is stopped. 3713 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3714 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3715 reset(); 3716 } 3717 } 3718} 3719 3720void AudioFlinger::PlaybackThread::Track::reset() 3721{ 3722 // Do not reset twice to avoid discarding data written just after a flush and before 3723 // the audioflinger thread detects the track is stopped. 3724 if (!mResetDone) { 3725 TrackBase::reset(); 3726 // Force underrun condition to avoid false underrun callback until first data is 3727 // written to buffer 3728 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3729 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3730 mFillingUpStatus = FS_FILLING; 3731 mResetDone = true; 3732 } 3733} 3734 3735void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3736{ 3737 mMute = muted; 3738} 3739 3740status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3741{ 3742 status_t status = DEAD_OBJECT; 3743 sp<ThreadBase> thread = mThread.promote(); 3744 if (thread != 0) { 3745 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3746 status = playbackThread->attachAuxEffect(this, EffectId); 3747 } 3748 return status; 3749} 3750 3751void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3752{ 3753 mAuxEffectId = EffectId; 3754 mAuxBuffer = buffer; 3755} 3756 3757// timed audio tracks 3758 3759sp<AudioFlinger::PlaybackThread::TimedTrack> 3760AudioFlinger::PlaybackThread::TimedTrack::create( 3761 PlaybackThread *thread, 3762 const sp<Client>& client, 3763 audio_stream_type_t streamType, 3764 uint32_t sampleRate, 3765 audio_format_t format, 3766 uint32_t channelMask, 3767 int frameCount, 3768 const sp<IMemory>& sharedBuffer, 3769 int sessionId) { 3770 if (!client->reserveTimedTrack()) 3771 return NULL; 3772 3773 sp<TimedTrack> track = new TimedTrack( 3774 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3775 sharedBuffer, sessionId); 3776 3777 if (track == NULL) { 3778 client->releaseTimedTrack(); 3779 return NULL; 3780 } 3781 3782 return track; 3783} 3784 3785AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3786 PlaybackThread *thread, 3787 const sp<Client>& client, 3788 audio_stream_type_t streamType, 3789 uint32_t sampleRate, 3790 audio_format_t format, 3791 uint32_t channelMask, 3792 int frameCount, 3793 const sp<IMemory>& sharedBuffer, 3794 int sessionId) 3795 : Track(thread, client, streamType, sampleRate, format, channelMask, 3796 frameCount, sharedBuffer, sessionId), 3797 mTimedSilenceBuffer(NULL), 3798 mTimedSilenceBufferSize(0), 3799 mTimedAudioOutputOnTime(false), 3800 mMediaTimeTransformValid(false) 3801{ 3802 LocalClock lc; 3803 mLocalTimeFreq = lc.getLocalFreq(); 3804 3805 mLocalTimeToSampleTransform.a_zero = 0; 3806 mLocalTimeToSampleTransform.b_zero = 0; 3807 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3808 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3809 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3810 &mLocalTimeToSampleTransform.a_to_b_denom); 3811} 3812 3813AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3814 mClient->releaseTimedTrack(); 3815 delete [] mTimedSilenceBuffer; 3816} 3817 3818status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3819 size_t size, sp<IMemory>* buffer) { 3820 3821 Mutex::Autolock _l(mTimedBufferQueueLock); 3822 3823 trimTimedBufferQueue_l(); 3824 3825 // lazily initialize the shared memory heap for timed buffers 3826 if (mTimedMemoryDealer == NULL) { 3827 const int kTimedBufferHeapSize = 512 << 10; 3828 3829 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3830 "AudioFlingerTimed"); 3831 if (mTimedMemoryDealer == NULL) 3832 return NO_MEMORY; 3833 } 3834 3835 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3836 if (newBuffer == NULL) { 3837 newBuffer = mTimedMemoryDealer->allocate(size); 3838 if (newBuffer == NULL) 3839 return NO_MEMORY; 3840 } 3841 3842 *buffer = newBuffer; 3843 return NO_ERROR; 3844} 3845 3846// caller must hold mTimedBufferQueueLock 3847void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3848 int64_t mediaTimeNow; 3849 { 3850 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3851 if (!mMediaTimeTransformValid) 3852 return; 3853 3854 int64_t targetTimeNow; 3855 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3856 ? mCCHelper.getCommonTime(&targetTimeNow) 3857 : mCCHelper.getLocalTime(&targetTimeNow); 3858 3859 if (OK != res) 3860 return; 3861 3862 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3863 &mediaTimeNow)) { 3864 return; 3865 } 3866 } 3867 3868 size_t trimIndex; 3869 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3870 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3871 break; 3872 } 3873 3874 if (trimIndex) { 3875 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3876 } 3877} 3878 3879status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3880 const sp<IMemory>& buffer, int64_t pts) { 3881 3882 { 3883 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3884 if (!mMediaTimeTransformValid) 3885 return INVALID_OPERATION; 3886 } 3887 3888 Mutex::Autolock _l(mTimedBufferQueueLock); 3889 3890 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3891 3892 return NO_ERROR; 3893} 3894 3895status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3896 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3897 3898 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3899 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3900 target); 3901 3902 if (!(target == TimedAudioTrack::LOCAL_TIME || 3903 target == TimedAudioTrack::COMMON_TIME)) { 3904 return BAD_VALUE; 3905 } 3906 3907 Mutex::Autolock lock(mMediaTimeTransformLock); 3908 mMediaTimeTransform = xform; 3909 mMediaTimeTransformTarget = target; 3910 mMediaTimeTransformValid = true; 3911 3912 return NO_ERROR; 3913} 3914 3915#define min(a, b) ((a) < (b) ? (a) : (b)) 3916 3917// implementation of getNextBuffer for tracks whose buffers have timestamps 3918status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3919 AudioBufferProvider::Buffer* buffer, int64_t pts) 3920{ 3921 if (pts == AudioBufferProvider::kInvalidPTS) { 3922 buffer->raw = 0; 3923 buffer->frameCount = 0; 3924 return INVALID_OPERATION; 3925 } 3926 3927 Mutex::Autolock _l(mTimedBufferQueueLock); 3928 3929 while (true) { 3930 3931 // if we have no timed buffers, then fail 3932 if (mTimedBufferQueue.isEmpty()) { 3933 buffer->raw = 0; 3934 buffer->frameCount = 0; 3935 return NOT_ENOUGH_DATA; 3936 } 3937 3938 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3939 3940 // calculate the PTS of the head of the timed buffer queue expressed in 3941 // local time 3942 int64_t headLocalPTS; 3943 { 3944 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3945 3946 assert(mMediaTimeTransformValid); 3947 3948 if (mMediaTimeTransform.a_to_b_denom == 0) { 3949 // the transform represents a pause, so yield silence 3950 timedYieldSilence(buffer->frameCount, buffer); 3951 return NO_ERROR; 3952 } 3953 3954 int64_t transformedPTS; 3955 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3956 &transformedPTS)) { 3957 // the transform failed. this shouldn't happen, but if it does 3958 // then just drop this buffer 3959 ALOGW("timedGetNextBuffer transform failed"); 3960 buffer->raw = 0; 3961 buffer->frameCount = 0; 3962 mTimedBufferQueue.removeAt(0); 3963 return NO_ERROR; 3964 } 3965 3966 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3967 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3968 &headLocalPTS)) { 3969 buffer->raw = 0; 3970 buffer->frameCount = 0; 3971 return INVALID_OPERATION; 3972 } 3973 } else { 3974 headLocalPTS = transformedPTS; 3975 } 3976 } 3977 3978 // adjust the head buffer's PTS to reflect the portion of the head buffer 3979 // that has already been consumed 3980 int64_t effectivePTS = headLocalPTS + 3981 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3982 3983 // Calculate the delta in samples between the head of the input buffer 3984 // queue and the start of the next output buffer that will be written. 3985 // If the transformation fails because of over or underflow, it means 3986 // that the sample's position in the output stream is so far out of 3987 // whack that it should just be dropped. 3988 int64_t sampleDelta; 3989 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3990 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3991 mTimedBufferQueue.removeAt(0); 3992 continue; 3993 } 3994 if (!mLocalTimeToSampleTransform.doForwardTransform( 3995 (effectivePTS - pts) << 32, &sampleDelta)) { 3996 ALOGV("*** too late during sample rate transform: dropped buffer"); 3997 mTimedBufferQueue.removeAt(0); 3998 continue; 3999 } 4000 4001 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4002 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4003 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4004 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4005 4006 // if the delta between the ideal placement for the next input sample and 4007 // the current output position is within this threshold, then we will 4008 // concatenate the next input samples to the previous output 4009 const int64_t kSampleContinuityThreshold = 4010 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4011 4012 // if this is the first buffer of audio that we're emitting from this track 4013 // then it should be almost exactly on time. 4014 const int64_t kSampleStartupThreshold = 1LL << 32; 4015 4016 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4017 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4018 // the next input is close enough to being on time, so concatenate it 4019 // with the last output 4020 timedYieldSamples(buffer); 4021 4022 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4023 return NO_ERROR; 4024 } else if (sampleDelta > 0) { 4025 // the gap between the current output position and the proper start of 4026 // the next input sample is too big, so fill it with silence 4027 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4028 4029 timedYieldSilence(framesUntilNextInput, buffer); 4030 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4031 return NO_ERROR; 4032 } else { 4033 // the next input sample is late 4034 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4035 size_t onTimeSamplePosition = 4036 head.position() + lateFrames * mCblk->frameSize; 4037 4038 if (onTimeSamplePosition > head.buffer()->size()) { 4039 // all the remaining samples in the head are too late, so 4040 // drop it and move on 4041 ALOGV("*** too late: dropped buffer"); 4042 mTimedBufferQueue.removeAt(0); 4043 continue; 4044 } else { 4045 // skip over the late samples 4046 head.setPosition(onTimeSamplePosition); 4047 4048 // yield the available samples 4049 timedYieldSamples(buffer); 4050 4051 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4052 return NO_ERROR; 4053 } 4054 } 4055 } 4056} 4057 4058// Yield samples from the timed buffer queue head up to the given output 4059// buffer's capacity. 4060// 4061// Caller must hold mTimedBufferQueueLock 4062void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4063 AudioBufferProvider::Buffer* buffer) { 4064 4065 const TimedBuffer& head = mTimedBufferQueue[0]; 4066 4067 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4068 head.position()); 4069 4070 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4071 mCblk->frameSize); 4072 size_t framesRequested = buffer->frameCount; 4073 buffer->frameCount = min(framesLeftInHead, framesRequested); 4074 4075 mTimedAudioOutputOnTime = true; 4076} 4077 4078// Yield samples of silence up to the given output buffer's capacity 4079// 4080// Caller must hold mTimedBufferQueueLock 4081void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4082 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4083 4084 // lazily allocate a buffer filled with silence 4085 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4086 delete [] mTimedSilenceBuffer; 4087 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4088 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4089 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4090 } 4091 4092 buffer->raw = mTimedSilenceBuffer; 4093 size_t framesRequested = buffer->frameCount; 4094 buffer->frameCount = min(numFrames, framesRequested); 4095 4096 mTimedAudioOutputOnTime = false; 4097} 4098 4099void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4100 AudioBufferProvider::Buffer* buffer) { 4101 4102 Mutex::Autolock _l(mTimedBufferQueueLock); 4103 4104 // If the buffer which was just released is part of the buffer at the head 4105 // of the queue, be sure to update the amt of the buffer which has been 4106 // consumed. If the buffer being returned is not part of the head of the 4107 // queue, its either because the buffer is part of the silence buffer, or 4108 // because the head of the timed queue was trimmed after the mixer called 4109 // getNextBuffer but before the mixer called releaseBuffer. 4110 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4111 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4112 4113 void* start = head.buffer()->pointer(); 4114 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4115 4116 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4117 head.setPosition(head.position() + 4118 (buffer->frameCount * mCblk->frameSize)); 4119 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4120 mTimedBufferQueue.removeAt(0); 4121 } 4122 } 4123 } 4124 4125 buffer->raw = 0; 4126 buffer->frameCount = 0; 4127} 4128 4129uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4130 Mutex::Autolock _l(mTimedBufferQueueLock); 4131 4132 uint32_t frames = 0; 4133 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4134 const TimedBuffer& tb = mTimedBufferQueue[i]; 4135 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4136 } 4137 4138 return frames; 4139} 4140 4141AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4142 : mPTS(0), mPosition(0) {} 4143 4144AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4145 const sp<IMemory>& buffer, int64_t pts) 4146 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4147 4148// ---------------------------------------------------------------------------- 4149 4150// RecordTrack constructor must be called with AudioFlinger::mLock held 4151AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4152 RecordThread *thread, 4153 const sp<Client>& client, 4154 uint32_t sampleRate, 4155 audio_format_t format, 4156 uint32_t channelMask, 4157 int frameCount, 4158 int sessionId) 4159 : TrackBase(thread, client, sampleRate, format, 4160 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4161 mOverflow(false) 4162{ 4163 if (mCblk != NULL) { 4164 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4165 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4166 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4167 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4168 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4169 } else { 4170 mCblk->frameSize = sizeof(int8_t); 4171 } 4172 } 4173} 4174 4175AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4176{ 4177 sp<ThreadBase> thread = mThread.promote(); 4178 if (thread != 0) { 4179 AudioSystem::releaseInput(thread->id()); 4180 } 4181} 4182 4183status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4184{ 4185 audio_track_cblk_t* cblk = this->cblk(); 4186 uint32_t framesAvail; 4187 uint32_t framesReq = buffer->frameCount; 4188 4189 // Check if last stepServer failed, try to step now 4190 if (mStepServerFailed) { 4191 if (!step()) goto getNextBuffer_exit; 4192 ALOGV("stepServer recovered"); 4193 mStepServerFailed = false; 4194 } 4195 4196 framesAvail = cblk->framesAvailable_l(); 4197 4198 if (CC_LIKELY(framesAvail)) { 4199 uint32_t s = cblk->server; 4200 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4201 4202 if (framesReq > framesAvail) { 4203 framesReq = framesAvail; 4204 } 4205 if (s + framesReq > bufferEnd) { 4206 framesReq = bufferEnd - s; 4207 } 4208 4209 buffer->raw = getBuffer(s, framesReq); 4210 if (buffer->raw == NULL) goto getNextBuffer_exit; 4211 4212 buffer->frameCount = framesReq; 4213 return NO_ERROR; 4214 } 4215 4216getNextBuffer_exit: 4217 buffer->raw = NULL; 4218 buffer->frameCount = 0; 4219 return NOT_ENOUGH_DATA; 4220} 4221 4222status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4223{ 4224 sp<ThreadBase> thread = mThread.promote(); 4225 if (thread != 0) { 4226 RecordThread *recordThread = (RecordThread *)thread.get(); 4227 return recordThread->start(this, tid); 4228 } else { 4229 return BAD_VALUE; 4230 } 4231} 4232 4233void AudioFlinger::RecordThread::RecordTrack::stop() 4234{ 4235 sp<ThreadBase> thread = mThread.promote(); 4236 if (thread != 0) { 4237 RecordThread *recordThread = (RecordThread *)thread.get(); 4238 recordThread->stop(this); 4239 TrackBase::reset(); 4240 // Force overerrun condition to avoid false overrun callback until first data is 4241 // read from buffer 4242 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4243 } 4244} 4245 4246void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4247{ 4248 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4249 (mClient == 0) ? getpid_cached : mClient->pid(), 4250 mFormat, 4251 mChannelMask, 4252 mSessionId, 4253 mFrameCount, 4254 mState, 4255 mCblk->sampleRate, 4256 mCblk->server, 4257 mCblk->user); 4258} 4259 4260 4261// ---------------------------------------------------------------------------- 4262 4263AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4264 PlaybackThread *playbackThread, 4265 DuplicatingThread *sourceThread, 4266 uint32_t sampleRate, 4267 audio_format_t format, 4268 uint32_t channelMask, 4269 int frameCount) 4270 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4271 mActive(false), mSourceThread(sourceThread) 4272{ 4273 4274 if (mCblk != NULL) { 4275 mCblk->flags |= CBLK_DIRECTION_OUT; 4276 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4277 mOutBuffer.frameCount = 0; 4278 playbackThread->mTracks.add(this); 4279 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4280 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4281 mCblk, mBuffer, mCblk->buffers, 4282 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4283 } else { 4284 ALOGW("Error creating output track on thread %p", playbackThread); 4285 } 4286} 4287 4288AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4289{ 4290 clearBufferQueue(); 4291} 4292 4293status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4294{ 4295 status_t status = Track::start(tid); 4296 if (status != NO_ERROR) { 4297 return status; 4298 } 4299 4300 mActive = true; 4301 mRetryCount = 127; 4302 return status; 4303} 4304 4305void AudioFlinger::PlaybackThread::OutputTrack::stop() 4306{ 4307 Track::stop(); 4308 clearBufferQueue(); 4309 mOutBuffer.frameCount = 0; 4310 mActive = false; 4311} 4312 4313bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4314{ 4315 Buffer *pInBuffer; 4316 Buffer inBuffer; 4317 uint32_t channelCount = mChannelCount; 4318 bool outputBufferFull = false; 4319 inBuffer.frameCount = frames; 4320 inBuffer.i16 = data; 4321 4322 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4323 4324 if (!mActive && frames != 0) { 4325 start(0); 4326 sp<ThreadBase> thread = mThread.promote(); 4327 if (thread != 0) { 4328 MixerThread *mixerThread = (MixerThread *)thread.get(); 4329 if (mCblk->frameCount > frames){ 4330 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4331 uint32_t startFrames = (mCblk->frameCount - frames); 4332 pInBuffer = new Buffer; 4333 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4334 pInBuffer->frameCount = startFrames; 4335 pInBuffer->i16 = pInBuffer->mBuffer; 4336 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4337 mBufferQueue.add(pInBuffer); 4338 } else { 4339 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4340 } 4341 } 4342 } 4343 } 4344 4345 while (waitTimeLeftMs) { 4346 // First write pending buffers, then new data 4347 if (mBufferQueue.size()) { 4348 pInBuffer = mBufferQueue.itemAt(0); 4349 } else { 4350 pInBuffer = &inBuffer; 4351 } 4352 4353 if (pInBuffer->frameCount == 0) { 4354 break; 4355 } 4356 4357 if (mOutBuffer.frameCount == 0) { 4358 mOutBuffer.frameCount = pInBuffer->frameCount; 4359 nsecs_t startTime = systemTime(); 4360 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4361 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4362 outputBufferFull = true; 4363 break; 4364 } 4365 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4366 if (waitTimeLeftMs >= waitTimeMs) { 4367 waitTimeLeftMs -= waitTimeMs; 4368 } else { 4369 waitTimeLeftMs = 0; 4370 } 4371 } 4372 4373 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4374 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4375 mCblk->stepUser(outFrames); 4376 pInBuffer->frameCount -= outFrames; 4377 pInBuffer->i16 += outFrames * channelCount; 4378 mOutBuffer.frameCount -= outFrames; 4379 mOutBuffer.i16 += outFrames * channelCount; 4380 4381 if (pInBuffer->frameCount == 0) { 4382 if (mBufferQueue.size()) { 4383 mBufferQueue.removeAt(0); 4384 delete [] pInBuffer->mBuffer; 4385 delete pInBuffer; 4386 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4387 } else { 4388 break; 4389 } 4390 } 4391 } 4392 4393 // If we could not write all frames, allocate a buffer and queue it for next time. 4394 if (inBuffer.frameCount) { 4395 sp<ThreadBase> thread = mThread.promote(); 4396 if (thread != 0 && !thread->standby()) { 4397 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4398 pInBuffer = new Buffer; 4399 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4400 pInBuffer->frameCount = inBuffer.frameCount; 4401 pInBuffer->i16 = pInBuffer->mBuffer; 4402 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4403 mBufferQueue.add(pInBuffer); 4404 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4405 } else { 4406 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4407 } 4408 } 4409 } 4410 4411 // Calling write() with a 0 length buffer, means that no more data will be written: 4412 // If no more buffers are pending, fill output track buffer to make sure it is started 4413 // by output mixer. 4414 if (frames == 0 && mBufferQueue.size() == 0) { 4415 if (mCblk->user < mCblk->frameCount) { 4416 frames = mCblk->frameCount - mCblk->user; 4417 pInBuffer = new Buffer; 4418 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4419 pInBuffer->frameCount = frames; 4420 pInBuffer->i16 = pInBuffer->mBuffer; 4421 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4422 mBufferQueue.add(pInBuffer); 4423 } else if (mActive) { 4424 stop(); 4425 } 4426 } 4427 4428 return outputBufferFull; 4429} 4430 4431status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4432{ 4433 int active; 4434 status_t result; 4435 audio_track_cblk_t* cblk = mCblk; 4436 uint32_t framesReq = buffer->frameCount; 4437 4438// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4439 buffer->frameCount = 0; 4440 4441 uint32_t framesAvail = cblk->framesAvailable(); 4442 4443 4444 if (framesAvail == 0) { 4445 Mutex::Autolock _l(cblk->lock); 4446 goto start_loop_here; 4447 while (framesAvail == 0) { 4448 active = mActive; 4449 if (CC_UNLIKELY(!active)) { 4450 ALOGV("Not active and NO_MORE_BUFFERS"); 4451 return NO_MORE_BUFFERS; 4452 } 4453 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4454 if (result != NO_ERROR) { 4455 return NO_MORE_BUFFERS; 4456 } 4457 // read the server count again 4458 start_loop_here: 4459 framesAvail = cblk->framesAvailable_l(); 4460 } 4461 } 4462 4463// if (framesAvail < framesReq) { 4464// return NO_MORE_BUFFERS; 4465// } 4466 4467 if (framesReq > framesAvail) { 4468 framesReq = framesAvail; 4469 } 4470 4471 uint32_t u = cblk->user; 4472 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4473 4474 if (u + framesReq > bufferEnd) { 4475 framesReq = bufferEnd - u; 4476 } 4477 4478 buffer->frameCount = framesReq; 4479 buffer->raw = (void *)cblk->buffer(u); 4480 return NO_ERROR; 4481} 4482 4483 4484void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4485{ 4486 size_t size = mBufferQueue.size(); 4487 4488 for (size_t i = 0; i < size; i++) { 4489 Buffer *pBuffer = mBufferQueue.itemAt(i); 4490 delete [] pBuffer->mBuffer; 4491 delete pBuffer; 4492 } 4493 mBufferQueue.clear(); 4494} 4495 4496// ---------------------------------------------------------------------------- 4497 4498AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4499 : RefBase(), 4500 mAudioFlinger(audioFlinger), 4501 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4502 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4503 mPid(pid), 4504 mTimedTrackCount(0) 4505{ 4506 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4507} 4508 4509// Client destructor must be called with AudioFlinger::mLock held 4510AudioFlinger::Client::~Client() 4511{ 4512 mAudioFlinger->removeClient_l(mPid); 4513} 4514 4515sp<MemoryDealer> AudioFlinger::Client::heap() const 4516{ 4517 return mMemoryDealer; 4518} 4519 4520// Reserve one of the limited slots for a timed audio track associated 4521// with this client 4522bool AudioFlinger::Client::reserveTimedTrack() 4523{ 4524 const int kMaxTimedTracksPerClient = 4; 4525 4526 Mutex::Autolock _l(mTimedTrackLock); 4527 4528 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4529 ALOGW("can not create timed track - pid %d has exceeded the limit", 4530 mPid); 4531 return false; 4532 } 4533 4534 mTimedTrackCount++; 4535 return true; 4536} 4537 4538// Release a slot for a timed audio track 4539void AudioFlinger::Client::releaseTimedTrack() 4540{ 4541 Mutex::Autolock _l(mTimedTrackLock); 4542 mTimedTrackCount--; 4543} 4544 4545// ---------------------------------------------------------------------------- 4546 4547AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4548 const sp<IAudioFlingerClient>& client, 4549 pid_t pid) 4550 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4551{ 4552} 4553 4554AudioFlinger::NotificationClient::~NotificationClient() 4555{ 4556} 4557 4558void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4559{ 4560 sp<NotificationClient> keep(this); 4561 mAudioFlinger->removeNotificationClient(mPid); 4562} 4563 4564// ---------------------------------------------------------------------------- 4565 4566AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4567 : BnAudioTrack(), 4568 mTrack(track) 4569{ 4570} 4571 4572AudioFlinger::TrackHandle::~TrackHandle() { 4573 // just stop the track on deletion, associated resources 4574 // will be freed from the main thread once all pending buffers have 4575 // been played. Unless it's not in the active track list, in which 4576 // case we free everything now... 4577 mTrack->destroy(); 4578} 4579 4580sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4581 return mTrack->getCblk(); 4582} 4583 4584status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4585 return mTrack->start(tid); 4586} 4587 4588void AudioFlinger::TrackHandle::stop() { 4589 mTrack->stop(); 4590} 4591 4592void AudioFlinger::TrackHandle::flush() { 4593 mTrack->flush(); 4594} 4595 4596void AudioFlinger::TrackHandle::mute(bool e) { 4597 mTrack->mute(e); 4598} 4599 4600void AudioFlinger::TrackHandle::pause() { 4601 mTrack->pause(); 4602} 4603 4604status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4605{ 4606 return mTrack->attachAuxEffect(EffectId); 4607} 4608 4609status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4610 sp<IMemory>* buffer) { 4611 if (!mTrack->isTimedTrack()) 4612 return INVALID_OPERATION; 4613 4614 PlaybackThread::TimedTrack* tt = 4615 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4616 return tt->allocateTimedBuffer(size, buffer); 4617} 4618 4619status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4620 int64_t pts) { 4621 if (!mTrack->isTimedTrack()) 4622 return INVALID_OPERATION; 4623 4624 PlaybackThread::TimedTrack* tt = 4625 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4626 return tt->queueTimedBuffer(buffer, pts); 4627} 4628 4629status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4630 const LinearTransform& xform, int target) { 4631 4632 if (!mTrack->isTimedTrack()) 4633 return INVALID_OPERATION; 4634 4635 PlaybackThread::TimedTrack* tt = 4636 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4637 return tt->setMediaTimeTransform( 4638 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4639} 4640 4641status_t AudioFlinger::TrackHandle::onTransact( 4642 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4643{ 4644 return BnAudioTrack::onTransact(code, data, reply, flags); 4645} 4646 4647// ---------------------------------------------------------------------------- 4648 4649sp<IAudioRecord> AudioFlinger::openRecord( 4650 pid_t pid, 4651 audio_io_handle_t input, 4652 uint32_t sampleRate, 4653 audio_format_t format, 4654 uint32_t channelMask, 4655 int frameCount, 4656 // FIXME dead, remove from IAudioFlinger 4657 uint32_t flags, 4658 int *sessionId, 4659 status_t *status) 4660{ 4661 sp<RecordThread::RecordTrack> recordTrack; 4662 sp<RecordHandle> recordHandle; 4663 sp<Client> client; 4664 status_t lStatus; 4665 RecordThread *thread; 4666 size_t inFrameCount; 4667 int lSessionId; 4668 4669 // check calling permissions 4670 if (!recordingAllowed()) { 4671 lStatus = PERMISSION_DENIED; 4672 goto Exit; 4673 } 4674 4675 // add client to list 4676 { // scope for mLock 4677 Mutex::Autolock _l(mLock); 4678 thread = checkRecordThread_l(input); 4679 if (thread == NULL) { 4680 lStatus = BAD_VALUE; 4681 goto Exit; 4682 } 4683 4684 client = registerPid_l(pid); 4685 4686 // If no audio session id is provided, create one here 4687 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4688 lSessionId = *sessionId; 4689 } else { 4690 lSessionId = nextUniqueId(); 4691 if (sessionId != NULL) { 4692 *sessionId = lSessionId; 4693 } 4694 } 4695 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4696 recordTrack = thread->createRecordTrack_l(client, 4697 sampleRate, 4698 format, 4699 channelMask, 4700 frameCount, 4701 lSessionId, 4702 &lStatus); 4703 } 4704 if (lStatus != NO_ERROR) { 4705 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4706 // destructor is called by the TrackBase destructor with mLock held 4707 client.clear(); 4708 recordTrack.clear(); 4709 goto Exit; 4710 } 4711 4712 // return to handle to client 4713 recordHandle = new RecordHandle(recordTrack); 4714 lStatus = NO_ERROR; 4715 4716Exit: 4717 if (status) { 4718 *status = lStatus; 4719 } 4720 return recordHandle; 4721} 4722 4723// ---------------------------------------------------------------------------- 4724 4725AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4726 : BnAudioRecord(), 4727 mRecordTrack(recordTrack) 4728{ 4729} 4730 4731AudioFlinger::RecordHandle::~RecordHandle() { 4732 stop(); 4733} 4734 4735sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4736 return mRecordTrack->getCblk(); 4737} 4738 4739status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4740 ALOGV("RecordHandle::start()"); 4741 return mRecordTrack->start(tid); 4742} 4743 4744void AudioFlinger::RecordHandle::stop() { 4745 ALOGV("RecordHandle::stop()"); 4746 mRecordTrack->stop(); 4747} 4748 4749status_t AudioFlinger::RecordHandle::onTransact( 4750 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4751{ 4752 return BnAudioRecord::onTransact(code, data, reply, flags); 4753} 4754 4755// ---------------------------------------------------------------------------- 4756 4757AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4758 AudioStreamIn *input, 4759 uint32_t sampleRate, 4760 uint32_t channels, 4761 audio_io_handle_t id, 4762 uint32_t device) : 4763 ThreadBase(audioFlinger, id, device, RECORD), 4764 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4765 // mRsmpInIndex and mInputBytes set by readInputParameters() 4766 mReqChannelCount(popcount(channels)), 4767 mReqSampleRate(sampleRate) 4768 // mBytesRead is only meaningful while active, and so is cleared in start() 4769 // (but might be better to also clear here for dump?) 4770{ 4771 snprintf(mName, kNameLength, "AudioIn_%d", id); 4772 4773 readInputParameters(); 4774} 4775 4776 4777AudioFlinger::RecordThread::~RecordThread() 4778{ 4779 delete[] mRsmpInBuffer; 4780 delete mResampler; 4781 delete[] mRsmpOutBuffer; 4782} 4783 4784void AudioFlinger::RecordThread::onFirstRef() 4785{ 4786 run(mName, PRIORITY_URGENT_AUDIO); 4787} 4788 4789status_t AudioFlinger::RecordThread::readyToRun() 4790{ 4791 status_t status = initCheck(); 4792 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4793 return status; 4794} 4795 4796bool AudioFlinger::RecordThread::threadLoop() 4797{ 4798 AudioBufferProvider::Buffer buffer; 4799 sp<RecordTrack> activeTrack; 4800 Vector< sp<EffectChain> > effectChains; 4801 4802 nsecs_t lastWarning = 0; 4803 4804 acquireWakeLock(); 4805 4806 // start recording 4807 while (!exitPending()) { 4808 4809 processConfigEvents(); 4810 4811 { // scope for mLock 4812 Mutex::Autolock _l(mLock); 4813 checkForNewParameters_l(); 4814 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4815 if (!mStandby) { 4816 mInput->stream->common.standby(&mInput->stream->common); 4817 mStandby = true; 4818 } 4819 4820 if (exitPending()) break; 4821 4822 releaseWakeLock_l(); 4823 ALOGV("RecordThread: loop stopping"); 4824 // go to sleep 4825 mWaitWorkCV.wait(mLock); 4826 ALOGV("RecordThread: loop starting"); 4827 acquireWakeLock_l(); 4828 continue; 4829 } 4830 if (mActiveTrack != 0) { 4831 if (mActiveTrack->mState == TrackBase::PAUSING) { 4832 if (!mStandby) { 4833 mInput->stream->common.standby(&mInput->stream->common); 4834 mStandby = true; 4835 } 4836 mActiveTrack.clear(); 4837 mStartStopCond.broadcast(); 4838 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4839 if (mReqChannelCount != mActiveTrack->channelCount()) { 4840 mActiveTrack.clear(); 4841 mStartStopCond.broadcast(); 4842 } else if (mBytesRead != 0) { 4843 // record start succeeds only if first read from audio input 4844 // succeeds 4845 if (mBytesRead > 0) { 4846 mActiveTrack->mState = TrackBase::ACTIVE; 4847 } else { 4848 mActiveTrack.clear(); 4849 } 4850 mStartStopCond.broadcast(); 4851 } 4852 mStandby = false; 4853 } 4854 } 4855 lockEffectChains_l(effectChains); 4856 } 4857 4858 if (mActiveTrack != 0) { 4859 if (mActiveTrack->mState != TrackBase::ACTIVE && 4860 mActiveTrack->mState != TrackBase::RESUMING) { 4861 unlockEffectChains(effectChains); 4862 usleep(kRecordThreadSleepUs); 4863 continue; 4864 } 4865 for (size_t i = 0; i < effectChains.size(); i ++) { 4866 effectChains[i]->process_l(); 4867 } 4868 4869 buffer.frameCount = mFrameCount; 4870 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4871 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4872 size_t framesOut = buffer.frameCount; 4873 if (mResampler == NULL) { 4874 // no resampling 4875 while (framesOut) { 4876 size_t framesIn = mFrameCount - mRsmpInIndex; 4877 if (framesIn) { 4878 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4879 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4880 if (framesIn > framesOut) 4881 framesIn = framesOut; 4882 mRsmpInIndex += framesIn; 4883 framesOut -= framesIn; 4884 if ((int)mChannelCount == mReqChannelCount || 4885 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4886 memcpy(dst, src, framesIn * mFrameSize); 4887 } else { 4888 int16_t *src16 = (int16_t *)src; 4889 int16_t *dst16 = (int16_t *)dst; 4890 if (mChannelCount == 1) { 4891 while (framesIn--) { 4892 *dst16++ = *src16; 4893 *dst16++ = *src16++; 4894 } 4895 } else { 4896 while (framesIn--) { 4897 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4898 src16 += 2; 4899 } 4900 } 4901 } 4902 } 4903 if (framesOut && mFrameCount == mRsmpInIndex) { 4904 if (framesOut == mFrameCount && 4905 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4906 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4907 framesOut = 0; 4908 } else { 4909 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4910 mRsmpInIndex = 0; 4911 } 4912 if (mBytesRead < 0) { 4913 ALOGE("Error reading audio input"); 4914 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4915 // Force input into standby so that it tries to 4916 // recover at next read attempt 4917 mInput->stream->common.standby(&mInput->stream->common); 4918 usleep(kRecordThreadSleepUs); 4919 } 4920 mRsmpInIndex = mFrameCount; 4921 framesOut = 0; 4922 buffer.frameCount = 0; 4923 } 4924 } 4925 } 4926 } else { 4927 // resampling 4928 4929 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4930 // alter output frame count as if we were expecting stereo samples 4931 if (mChannelCount == 1 && mReqChannelCount == 1) { 4932 framesOut >>= 1; 4933 } 4934 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4935 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4936 // are 32 bit aligned which should be always true. 4937 if (mChannelCount == 2 && mReqChannelCount == 1) { 4938 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4939 // the resampler always outputs stereo samples: do post stereo to mono conversion 4940 int16_t *src = (int16_t *)mRsmpOutBuffer; 4941 int16_t *dst = buffer.i16; 4942 while (framesOut--) { 4943 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4944 src += 2; 4945 } 4946 } else { 4947 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4948 } 4949 4950 } 4951 mActiveTrack->releaseBuffer(&buffer); 4952 mActiveTrack->overflow(); 4953 } 4954 // client isn't retrieving buffers fast enough 4955 else { 4956 if (!mActiveTrack->setOverflow()) { 4957 nsecs_t now = systemTime(); 4958 if ((now - lastWarning) > kWarningThrottleNs) { 4959 ALOGW("RecordThread: buffer overflow"); 4960 lastWarning = now; 4961 } 4962 } 4963 // Release the processor for a while before asking for a new buffer. 4964 // This will give the application more chance to read from the buffer and 4965 // clear the overflow. 4966 usleep(kRecordThreadSleepUs); 4967 } 4968 } 4969 // enable changes in effect chain 4970 unlockEffectChains(effectChains); 4971 effectChains.clear(); 4972 } 4973 4974 if (!mStandby) { 4975 mInput->stream->common.standby(&mInput->stream->common); 4976 } 4977 mActiveTrack.clear(); 4978 4979 mStartStopCond.broadcast(); 4980 4981 releaseWakeLock(); 4982 4983 ALOGV("RecordThread %p exiting", this); 4984 return false; 4985} 4986 4987 4988sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4989 const sp<AudioFlinger::Client>& client, 4990 uint32_t sampleRate, 4991 audio_format_t format, 4992 int channelMask, 4993 int frameCount, 4994 int sessionId, 4995 status_t *status) 4996{ 4997 sp<RecordTrack> track; 4998 status_t lStatus; 4999 5000 lStatus = initCheck(); 5001 if (lStatus != NO_ERROR) { 5002 ALOGE("Audio driver not initialized."); 5003 goto Exit; 5004 } 5005 5006 { // scope for mLock 5007 Mutex::Autolock _l(mLock); 5008 5009 track = new RecordTrack(this, client, sampleRate, 5010 format, channelMask, frameCount, sessionId); 5011 5012 if (track->getCblk() == 0) { 5013 lStatus = NO_MEMORY; 5014 goto Exit; 5015 } 5016 5017 mTrack = track.get(); 5018 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5019 bool suspend = audio_is_bluetooth_sco_device( 5020 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5021 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5022 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5023 } 5024 lStatus = NO_ERROR; 5025 5026Exit: 5027 if (status) { 5028 *status = lStatus; 5029 } 5030 return track; 5031} 5032 5033status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5034{ 5035 ALOGV("RecordThread::start tid=%d", tid); 5036 sp <ThreadBase> strongMe = this; 5037 status_t status = NO_ERROR; 5038 { 5039 AutoMutex lock(mLock); 5040 if (mActiveTrack != 0) { 5041 if (recordTrack != mActiveTrack.get()) { 5042 status = -EBUSY; 5043 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5044 mActiveTrack->mState = TrackBase::ACTIVE; 5045 } 5046 return status; 5047 } 5048 5049 recordTrack->mState = TrackBase::IDLE; 5050 mActiveTrack = recordTrack; 5051 mLock.unlock(); 5052 status_t status = AudioSystem::startInput(mId); 5053 mLock.lock(); 5054 if (status != NO_ERROR) { 5055 mActiveTrack.clear(); 5056 return status; 5057 } 5058 mRsmpInIndex = mFrameCount; 5059 mBytesRead = 0; 5060 if (mResampler != NULL) { 5061 mResampler->reset(); 5062 } 5063 mActiveTrack->mState = TrackBase::RESUMING; 5064 // signal thread to start 5065 ALOGV("Signal record thread"); 5066 mWaitWorkCV.signal(); 5067 // do not wait for mStartStopCond if exiting 5068 if (exitPending()) { 5069 mActiveTrack.clear(); 5070 status = INVALID_OPERATION; 5071 goto startError; 5072 } 5073 mStartStopCond.wait(mLock); 5074 if (mActiveTrack == 0) { 5075 ALOGV("Record failed to start"); 5076 status = BAD_VALUE; 5077 goto startError; 5078 } 5079 ALOGV("Record started OK"); 5080 return status; 5081 } 5082startError: 5083 AudioSystem::stopInput(mId); 5084 return status; 5085} 5086 5087void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5088 ALOGV("RecordThread::stop"); 5089 sp <ThreadBase> strongMe = this; 5090 { 5091 AutoMutex lock(mLock); 5092 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5093 mActiveTrack->mState = TrackBase::PAUSING; 5094 // do not wait for mStartStopCond if exiting 5095 if (exitPending()) { 5096 return; 5097 } 5098 mStartStopCond.wait(mLock); 5099 // if we have been restarted, recordTrack == mActiveTrack.get() here 5100 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5101 mLock.unlock(); 5102 AudioSystem::stopInput(mId); 5103 mLock.lock(); 5104 ALOGV("Record stopped OK"); 5105 } 5106 } 5107 } 5108} 5109 5110status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5111{ 5112 const size_t SIZE = 256; 5113 char buffer[SIZE]; 5114 String8 result; 5115 5116 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5117 result.append(buffer); 5118 5119 if (mActiveTrack != 0) { 5120 result.append("Active Track:\n"); 5121 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5122 mActiveTrack->dump(buffer, SIZE); 5123 result.append(buffer); 5124 5125 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5126 result.append(buffer); 5127 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5128 result.append(buffer); 5129 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5130 result.append(buffer); 5131 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5132 result.append(buffer); 5133 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5134 result.append(buffer); 5135 5136 5137 } else { 5138 result.append("No record client\n"); 5139 } 5140 write(fd, result.string(), result.size()); 5141 5142 dumpBase(fd, args); 5143 dumpEffectChains(fd, args); 5144 5145 return NO_ERROR; 5146} 5147 5148status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5149{ 5150 size_t framesReq = buffer->frameCount; 5151 size_t framesReady = mFrameCount - mRsmpInIndex; 5152 int channelCount; 5153 5154 if (framesReady == 0) { 5155 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5156 if (mBytesRead < 0) { 5157 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5158 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5159 // Force input into standby so that it tries to 5160 // recover at next read attempt 5161 mInput->stream->common.standby(&mInput->stream->common); 5162 usleep(kRecordThreadSleepUs); 5163 } 5164 buffer->raw = NULL; 5165 buffer->frameCount = 0; 5166 return NOT_ENOUGH_DATA; 5167 } 5168 mRsmpInIndex = 0; 5169 framesReady = mFrameCount; 5170 } 5171 5172 if (framesReq > framesReady) { 5173 framesReq = framesReady; 5174 } 5175 5176 if (mChannelCount == 1 && mReqChannelCount == 2) { 5177 channelCount = 1; 5178 } else { 5179 channelCount = 2; 5180 } 5181 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5182 buffer->frameCount = framesReq; 5183 return NO_ERROR; 5184} 5185 5186void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5187{ 5188 mRsmpInIndex += buffer->frameCount; 5189 buffer->frameCount = 0; 5190} 5191 5192bool AudioFlinger::RecordThread::checkForNewParameters_l() 5193{ 5194 bool reconfig = false; 5195 5196 while (!mNewParameters.isEmpty()) { 5197 status_t status = NO_ERROR; 5198 String8 keyValuePair = mNewParameters[0]; 5199 AudioParameter param = AudioParameter(keyValuePair); 5200 int value; 5201 audio_format_t reqFormat = mFormat; 5202 int reqSamplingRate = mReqSampleRate; 5203 int reqChannelCount = mReqChannelCount; 5204 5205 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5206 reqSamplingRate = value; 5207 reconfig = true; 5208 } 5209 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5210 reqFormat = (audio_format_t) value; 5211 reconfig = true; 5212 } 5213 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5214 reqChannelCount = popcount(value); 5215 reconfig = true; 5216 } 5217 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5218 // do not accept frame count changes if tracks are open as the track buffer 5219 // size depends on frame count and correct behavior would not be guaranteed 5220 // if frame count is changed after track creation 5221 if (mActiveTrack != 0) { 5222 status = INVALID_OPERATION; 5223 } else { 5224 reconfig = true; 5225 } 5226 } 5227 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5228 // forward device change to effects that have requested to be 5229 // aware of attached audio device. 5230 for (size_t i = 0; i < mEffectChains.size(); i++) { 5231 mEffectChains[i]->setDevice_l(value); 5232 } 5233 // store input device and output device but do not forward output device to audio HAL. 5234 // Note that status is ignored by the caller for output device 5235 // (see AudioFlinger::setParameters() 5236 if (value & AUDIO_DEVICE_OUT_ALL) { 5237 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5238 status = BAD_VALUE; 5239 } else { 5240 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5241 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5242 if (mTrack != NULL) { 5243 bool suspend = audio_is_bluetooth_sco_device( 5244 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5245 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5246 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5247 } 5248 } 5249 mDevice |= (uint32_t)value; 5250 } 5251 if (status == NO_ERROR) { 5252 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5253 if (status == INVALID_OPERATION) { 5254 mInput->stream->common.standby(&mInput->stream->common); 5255 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5256 } 5257 if (reconfig) { 5258 if (status == BAD_VALUE && 5259 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5260 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5261 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5262 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5263 (reqChannelCount < 3)) { 5264 status = NO_ERROR; 5265 } 5266 if (status == NO_ERROR) { 5267 readInputParameters(); 5268 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5269 } 5270 } 5271 } 5272 5273 mNewParameters.removeAt(0); 5274 5275 mParamStatus = status; 5276 mParamCond.signal(); 5277 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5278 // already timed out waiting for the status and will never signal the condition. 5279 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5280 } 5281 return reconfig; 5282} 5283 5284String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5285{ 5286 char *s; 5287 String8 out_s8 = String8(); 5288 5289 Mutex::Autolock _l(mLock); 5290 if (initCheck() != NO_ERROR) { 5291 return out_s8; 5292 } 5293 5294 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5295 out_s8 = String8(s); 5296 free(s); 5297 return out_s8; 5298} 5299 5300void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5301 AudioSystem::OutputDescriptor desc; 5302 void *param2 = NULL; 5303 5304 switch (event) { 5305 case AudioSystem::INPUT_OPENED: 5306 case AudioSystem::INPUT_CONFIG_CHANGED: 5307 desc.channels = mChannelMask; 5308 desc.samplingRate = mSampleRate; 5309 desc.format = mFormat; 5310 desc.frameCount = mFrameCount; 5311 desc.latency = 0; 5312 param2 = &desc; 5313 break; 5314 5315 case AudioSystem::INPUT_CLOSED: 5316 default: 5317 break; 5318 } 5319 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5320} 5321 5322void AudioFlinger::RecordThread::readInputParameters() 5323{ 5324 delete mRsmpInBuffer; 5325 // mRsmpInBuffer is always assigned a new[] below 5326 delete mRsmpOutBuffer; 5327 mRsmpOutBuffer = NULL; 5328 delete mResampler; 5329 mResampler = NULL; 5330 5331 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5332 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5333 mChannelCount = (uint16_t)popcount(mChannelMask); 5334 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5335 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5336 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5337 mFrameCount = mInputBytes / mFrameSize; 5338 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5339 5340 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5341 { 5342 int channelCount; 5343 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5344 // stereo to mono post process as the resampler always outputs stereo. 5345 if (mChannelCount == 1 && mReqChannelCount == 2) { 5346 channelCount = 1; 5347 } else { 5348 channelCount = 2; 5349 } 5350 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5351 mResampler->setSampleRate(mSampleRate); 5352 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5353 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5354 5355 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5356 if (mChannelCount == 1 && mReqChannelCount == 1) { 5357 mFrameCount >>= 1; 5358 } 5359 5360 } 5361 mRsmpInIndex = mFrameCount; 5362} 5363 5364unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5365{ 5366 Mutex::Autolock _l(mLock); 5367 if (initCheck() != NO_ERROR) { 5368 return 0; 5369 } 5370 5371 return mInput->stream->get_input_frames_lost(mInput->stream); 5372} 5373 5374uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5375{ 5376 Mutex::Autolock _l(mLock); 5377 uint32_t result = 0; 5378 if (getEffectChain_l(sessionId) != 0) { 5379 result = EFFECT_SESSION; 5380 } 5381 5382 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5383 result |= TRACK_SESSION; 5384 } 5385 5386 return result; 5387} 5388 5389AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5390{ 5391 Mutex::Autolock _l(mLock); 5392 return mTrack; 5393} 5394 5395AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5396{ 5397 Mutex::Autolock _l(mLock); 5398 return mInput; 5399} 5400 5401AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5402{ 5403 Mutex::Autolock _l(mLock); 5404 AudioStreamIn *input = mInput; 5405 mInput = NULL; 5406 return input; 5407} 5408 5409// this method must always be called either with ThreadBase mLock held or inside the thread loop 5410audio_stream_t* AudioFlinger::RecordThread::stream() 5411{ 5412 if (mInput == NULL) { 5413 return NULL; 5414 } 5415 return &mInput->stream->common; 5416} 5417 5418 5419// ---------------------------------------------------------------------------- 5420 5421audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5422 uint32_t *pSamplingRate, 5423 audio_format_t *pFormat, 5424 uint32_t *pChannels, 5425 uint32_t *pLatencyMs, 5426 uint32_t flags) 5427{ 5428 status_t status; 5429 PlaybackThread *thread = NULL; 5430 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5431 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5432 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5433 uint32_t channels = pChannels ? *pChannels : 0; 5434 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5435 audio_stream_out_t *outStream; 5436 audio_hw_device_t *outHwDev; 5437 5438 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5439 pDevices ? *pDevices : 0, 5440 samplingRate, 5441 format, 5442 channels, 5443 flags); 5444 5445 if (pDevices == NULL || *pDevices == 0) { 5446 return 0; 5447 } 5448 5449 Mutex::Autolock _l(mLock); 5450 5451 outHwDev = findSuitableHwDev_l(*pDevices); 5452 if (outHwDev == NULL) 5453 return 0; 5454 5455 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5456 &channels, &samplingRate, &outStream); 5457 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5458 outStream, 5459 samplingRate, 5460 format, 5461 channels, 5462 status); 5463 5464 mHardwareStatus = AUDIO_HW_IDLE; 5465 if (outStream != NULL) { 5466 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5467 audio_io_handle_t id = nextUniqueId(); 5468 5469 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5470 (format != AUDIO_FORMAT_PCM_16_BIT) || 5471 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5472 thread = new DirectOutputThread(this, output, id, *pDevices); 5473 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5474 } else { 5475 thread = new MixerThread(this, output, id, *pDevices); 5476 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5477 } 5478 mPlaybackThreads.add(id, thread); 5479 5480 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5481 if (pFormat != NULL) *pFormat = format; 5482 if (pChannels != NULL) *pChannels = channels; 5483 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5484 5485 // notify client processes of the new output creation 5486 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5487 return id; 5488 } 5489 5490 return 0; 5491} 5492 5493audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5494 audio_io_handle_t output2) 5495{ 5496 Mutex::Autolock _l(mLock); 5497 MixerThread *thread1 = checkMixerThread_l(output1); 5498 MixerThread *thread2 = checkMixerThread_l(output2); 5499 5500 if (thread1 == NULL || thread2 == NULL) { 5501 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5502 return 0; 5503 } 5504 5505 audio_io_handle_t id = nextUniqueId(); 5506 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5507 thread->addOutputTrack(thread2); 5508 mPlaybackThreads.add(id, thread); 5509 // notify client processes of the new output creation 5510 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5511 return id; 5512} 5513 5514status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5515{ 5516 // keep strong reference on the playback thread so that 5517 // it is not destroyed while exit() is executed 5518 sp <PlaybackThread> thread; 5519 { 5520 Mutex::Autolock _l(mLock); 5521 thread = checkPlaybackThread_l(output); 5522 if (thread == NULL) { 5523 return BAD_VALUE; 5524 } 5525 5526 ALOGV("closeOutput() %d", output); 5527 5528 if (thread->type() == ThreadBase::MIXER) { 5529 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5530 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5531 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5532 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5533 } 5534 } 5535 } 5536 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5537 mPlaybackThreads.removeItem(output); 5538 } 5539 thread->exit(); 5540 // The thread entity (active unit of execution) is no longer running here, 5541 // but the ThreadBase container still exists. 5542 5543 if (thread->type() != ThreadBase::DUPLICATING) { 5544 AudioStreamOut *out = thread->clearOutput(); 5545 assert(out != NULL); 5546 // from now on thread->mOutput is NULL 5547 out->hwDev->close_output_stream(out->hwDev, out->stream); 5548 delete out; 5549 } 5550 return NO_ERROR; 5551} 5552 5553status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5554{ 5555 Mutex::Autolock _l(mLock); 5556 PlaybackThread *thread = checkPlaybackThread_l(output); 5557 5558 if (thread == NULL) { 5559 return BAD_VALUE; 5560 } 5561 5562 ALOGV("suspendOutput() %d", output); 5563 thread->suspend(); 5564 5565 return NO_ERROR; 5566} 5567 5568status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5569{ 5570 Mutex::Autolock _l(mLock); 5571 PlaybackThread *thread = checkPlaybackThread_l(output); 5572 5573 if (thread == NULL) { 5574 return BAD_VALUE; 5575 } 5576 5577 ALOGV("restoreOutput() %d", output); 5578 5579 thread->restore(); 5580 5581 return NO_ERROR; 5582} 5583 5584audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5585 uint32_t *pSamplingRate, 5586 audio_format_t *pFormat, 5587 uint32_t *pChannels, 5588 audio_in_acoustics_t acoustics) 5589{ 5590 status_t status; 5591 RecordThread *thread = NULL; 5592 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5593 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5594 uint32_t channels = pChannels ? *pChannels : 0; 5595 uint32_t reqSamplingRate = samplingRate; 5596 audio_format_t reqFormat = format; 5597 uint32_t reqChannels = channels; 5598 audio_stream_in_t *inStream; 5599 audio_hw_device_t *inHwDev; 5600 5601 if (pDevices == NULL || *pDevices == 0) { 5602 return 0; 5603 } 5604 5605 Mutex::Autolock _l(mLock); 5606 5607 inHwDev = findSuitableHwDev_l(*pDevices); 5608 if (inHwDev == NULL) 5609 return 0; 5610 5611 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5612 &channels, &samplingRate, 5613 acoustics, 5614 &inStream); 5615 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5616 inStream, 5617 samplingRate, 5618 format, 5619 channels, 5620 acoustics, 5621 status); 5622 5623 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5624 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5625 // or stereo to mono conversions on 16 bit PCM inputs. 5626 if (inStream == NULL && status == BAD_VALUE && 5627 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5628 (samplingRate <= 2 * reqSamplingRate) && 5629 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5630 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5631 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5632 &channels, &samplingRate, 5633 acoustics, 5634 &inStream); 5635 } 5636 5637 if (inStream != NULL) { 5638 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5639 5640 audio_io_handle_t id = nextUniqueId(); 5641 // Start record thread 5642 // RecorThread require both input and output device indication to forward to audio 5643 // pre processing modules 5644 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5645 thread = new RecordThread(this, 5646 input, 5647 reqSamplingRate, 5648 reqChannels, 5649 id, 5650 device); 5651 mRecordThreads.add(id, thread); 5652 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5653 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5654 if (pFormat != NULL) *pFormat = format; 5655 if (pChannels != NULL) *pChannels = reqChannels; 5656 5657 input->stream->common.standby(&input->stream->common); 5658 5659 // notify client processes of the new input creation 5660 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5661 return id; 5662 } 5663 5664 return 0; 5665} 5666 5667status_t AudioFlinger::closeInput(audio_io_handle_t input) 5668{ 5669 // keep strong reference on the record thread so that 5670 // it is not destroyed while exit() is executed 5671 sp <RecordThread> thread; 5672 { 5673 Mutex::Autolock _l(mLock); 5674 thread = checkRecordThread_l(input); 5675 if (thread == NULL) { 5676 return BAD_VALUE; 5677 } 5678 5679 ALOGV("closeInput() %d", input); 5680 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5681 mRecordThreads.removeItem(input); 5682 } 5683 thread->exit(); 5684 // The thread entity (active unit of execution) is no longer running here, 5685 // but the ThreadBase container still exists. 5686 5687 AudioStreamIn *in = thread->clearInput(); 5688 assert(in != NULL); 5689 // from now on thread->mInput is NULL 5690 in->hwDev->close_input_stream(in->hwDev, in->stream); 5691 delete in; 5692 5693 return NO_ERROR; 5694} 5695 5696status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5697{ 5698 Mutex::Autolock _l(mLock); 5699 MixerThread *dstThread = checkMixerThread_l(output); 5700 if (dstThread == NULL) { 5701 ALOGW("setStreamOutput() bad output id %d", output); 5702 return BAD_VALUE; 5703 } 5704 5705 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5706 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5707 5708 dstThread->setStreamValid(stream, true); 5709 5710 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5711 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5712 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5713 MixerThread *srcThread = (MixerThread *)thread; 5714 srcThread->setStreamValid(stream, false); 5715 srcThread->invalidateTracks(stream); 5716 } 5717 } 5718 5719 return NO_ERROR; 5720} 5721 5722 5723int AudioFlinger::newAudioSessionId() 5724{ 5725 return nextUniqueId(); 5726} 5727 5728void AudioFlinger::acquireAudioSessionId(int audioSession) 5729{ 5730 Mutex::Autolock _l(mLock); 5731 pid_t caller = IPCThreadState::self()->getCallingPid(); 5732 ALOGV("acquiring %d from %d", audioSession, caller); 5733 size_t num = mAudioSessionRefs.size(); 5734 for (size_t i = 0; i< num; i++) { 5735 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5736 if (ref->sessionid == audioSession && ref->pid == caller) { 5737 ref->cnt++; 5738 ALOGV(" incremented refcount to %d", ref->cnt); 5739 return; 5740 } 5741 } 5742 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5743 ALOGV(" added new entry for %d", audioSession); 5744} 5745 5746void AudioFlinger::releaseAudioSessionId(int audioSession) 5747{ 5748 Mutex::Autolock _l(mLock); 5749 pid_t caller = IPCThreadState::self()->getCallingPid(); 5750 ALOGV("releasing %d from %d", audioSession, caller); 5751 size_t num = mAudioSessionRefs.size(); 5752 for (size_t i = 0; i< num; i++) { 5753 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5754 if (ref->sessionid == audioSession && ref->pid == caller) { 5755 ref->cnt--; 5756 ALOGV(" decremented refcount to %d", ref->cnt); 5757 if (ref->cnt == 0) { 5758 mAudioSessionRefs.removeAt(i); 5759 delete ref; 5760 purgeStaleEffects_l(); 5761 } 5762 return; 5763 } 5764 } 5765 ALOGW("session id %d not found for pid %d", audioSession, caller); 5766} 5767 5768void AudioFlinger::purgeStaleEffects_l() { 5769 5770 ALOGV("purging stale effects"); 5771 5772 Vector< sp<EffectChain> > chains; 5773 5774 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5775 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5776 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5777 sp<EffectChain> ec = t->mEffectChains[j]; 5778 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5779 chains.push(ec); 5780 } 5781 } 5782 } 5783 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5784 sp<RecordThread> t = mRecordThreads.valueAt(i); 5785 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5786 sp<EffectChain> ec = t->mEffectChains[j]; 5787 chains.push(ec); 5788 } 5789 } 5790 5791 for (size_t i = 0; i < chains.size(); i++) { 5792 sp<EffectChain> ec = chains[i]; 5793 int sessionid = ec->sessionId(); 5794 sp<ThreadBase> t = ec->mThread.promote(); 5795 if (t == 0) { 5796 continue; 5797 } 5798 size_t numsessionrefs = mAudioSessionRefs.size(); 5799 bool found = false; 5800 for (size_t k = 0; k < numsessionrefs; k++) { 5801 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5802 if (ref->sessionid == sessionid) { 5803 ALOGV(" session %d still exists for %d with %d refs", 5804 sessionid, ref->pid, ref->cnt); 5805 found = true; 5806 break; 5807 } 5808 } 5809 if (!found) { 5810 // remove all effects from the chain 5811 while (ec->mEffects.size()) { 5812 sp<EffectModule> effect = ec->mEffects[0]; 5813 effect->unPin(); 5814 Mutex::Autolock _l (t->mLock); 5815 t->removeEffect_l(effect); 5816 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5817 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5818 if (handle != 0) { 5819 handle->mEffect.clear(); 5820 if (handle->mHasControl && handle->mEnabled) { 5821 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5822 } 5823 } 5824 } 5825 AudioSystem::unregisterEffect(effect->id()); 5826 } 5827 } 5828 } 5829 return; 5830} 5831 5832// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5833AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5834{ 5835 return mPlaybackThreads.valueFor(output).get(); 5836} 5837 5838// checkMixerThread_l() must be called with AudioFlinger::mLock held 5839AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5840{ 5841 PlaybackThread *thread = checkPlaybackThread_l(output); 5842 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5843} 5844 5845// checkRecordThread_l() must be called with AudioFlinger::mLock held 5846AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5847{ 5848 return mRecordThreads.valueFor(input).get(); 5849} 5850 5851uint32_t AudioFlinger::nextUniqueId() 5852{ 5853 return android_atomic_inc(&mNextUniqueId); 5854} 5855 5856AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5857{ 5858 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5859 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5860 AudioStreamOut *output = thread->getOutput(); 5861 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5862 return thread; 5863 } 5864 } 5865 return NULL; 5866} 5867 5868uint32_t AudioFlinger::primaryOutputDevice_l() 5869{ 5870 PlaybackThread *thread = primaryPlaybackThread_l(); 5871 5872 if (thread == NULL) { 5873 return 0; 5874 } 5875 5876 return thread->device(); 5877} 5878 5879 5880// ---------------------------------------------------------------------------- 5881// Effect management 5882// ---------------------------------------------------------------------------- 5883 5884 5885status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5886{ 5887 Mutex::Autolock _l(mLock); 5888 return EffectQueryNumberEffects(numEffects); 5889} 5890 5891status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5892{ 5893 Mutex::Autolock _l(mLock); 5894 return EffectQueryEffect(index, descriptor); 5895} 5896 5897status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5898 effect_descriptor_t *descriptor) const 5899{ 5900 Mutex::Autolock _l(mLock); 5901 return EffectGetDescriptor(pUuid, descriptor); 5902} 5903 5904 5905sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5906 effect_descriptor_t *pDesc, 5907 const sp<IEffectClient>& effectClient, 5908 int32_t priority, 5909 audio_io_handle_t io, 5910 int sessionId, 5911 status_t *status, 5912 int *id, 5913 int *enabled) 5914{ 5915 status_t lStatus = NO_ERROR; 5916 sp<EffectHandle> handle; 5917 effect_descriptor_t desc; 5918 5919 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5920 pid, effectClient.get(), priority, sessionId, io); 5921 5922 if (pDesc == NULL) { 5923 lStatus = BAD_VALUE; 5924 goto Exit; 5925 } 5926 5927 // check audio settings permission for global effects 5928 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5929 lStatus = PERMISSION_DENIED; 5930 goto Exit; 5931 } 5932 5933 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5934 // that can only be created by audio policy manager (running in same process) 5935 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5936 lStatus = PERMISSION_DENIED; 5937 goto Exit; 5938 } 5939 5940 if (io == 0) { 5941 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5942 // output must be specified by AudioPolicyManager when using session 5943 // AUDIO_SESSION_OUTPUT_STAGE 5944 lStatus = BAD_VALUE; 5945 goto Exit; 5946 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5947 // if the output returned by getOutputForEffect() is removed before we lock the 5948 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5949 // and we will exit safely 5950 io = AudioSystem::getOutputForEffect(&desc); 5951 } 5952 } 5953 5954 { 5955 Mutex::Autolock _l(mLock); 5956 5957 5958 if (!EffectIsNullUuid(&pDesc->uuid)) { 5959 // if uuid is specified, request effect descriptor 5960 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5961 if (lStatus < 0) { 5962 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5963 goto Exit; 5964 } 5965 } else { 5966 // if uuid is not specified, look for an available implementation 5967 // of the required type in effect factory 5968 if (EffectIsNullUuid(&pDesc->type)) { 5969 ALOGW("createEffect() no effect type"); 5970 lStatus = BAD_VALUE; 5971 goto Exit; 5972 } 5973 uint32_t numEffects = 0; 5974 effect_descriptor_t d; 5975 d.flags = 0; // prevent compiler warning 5976 bool found = false; 5977 5978 lStatus = EffectQueryNumberEffects(&numEffects); 5979 if (lStatus < 0) { 5980 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5981 goto Exit; 5982 } 5983 for (uint32_t i = 0; i < numEffects; i++) { 5984 lStatus = EffectQueryEffect(i, &desc); 5985 if (lStatus < 0) { 5986 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5987 continue; 5988 } 5989 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5990 // If matching type found save effect descriptor. If the session is 5991 // 0 and the effect is not auxiliary, continue enumeration in case 5992 // an auxiliary version of this effect type is available 5993 found = true; 5994 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5995 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5996 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5997 break; 5998 } 5999 } 6000 } 6001 if (!found) { 6002 lStatus = BAD_VALUE; 6003 ALOGW("createEffect() effect not found"); 6004 goto Exit; 6005 } 6006 // For same effect type, chose auxiliary version over insert version if 6007 // connect to output mix (Compliance to OpenSL ES) 6008 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6009 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6010 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6011 } 6012 } 6013 6014 // Do not allow auxiliary effects on a session different from 0 (output mix) 6015 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6016 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6017 lStatus = INVALID_OPERATION; 6018 goto Exit; 6019 } 6020 6021 // check recording permission for visualizer 6022 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6023 !recordingAllowed()) { 6024 lStatus = PERMISSION_DENIED; 6025 goto Exit; 6026 } 6027 6028 // return effect descriptor 6029 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6030 6031 // If output is not specified try to find a matching audio session ID in one of the 6032 // output threads. 6033 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6034 // because of code checking output when entering the function. 6035 // Note: io is never 0 when creating an effect on an input 6036 if (io == 0) { 6037 // look for the thread where the specified audio session is present 6038 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6039 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6040 io = mPlaybackThreads.keyAt(i); 6041 break; 6042 } 6043 } 6044 if (io == 0) { 6045 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6046 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6047 io = mRecordThreads.keyAt(i); 6048 break; 6049 } 6050 } 6051 } 6052 // If no output thread contains the requested session ID, default to 6053 // first output. The effect chain will be moved to the correct output 6054 // thread when a track with the same session ID is created 6055 if (io == 0 && mPlaybackThreads.size()) { 6056 io = mPlaybackThreads.keyAt(0); 6057 } 6058 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6059 } 6060 ThreadBase *thread = checkRecordThread_l(io); 6061 if (thread == NULL) { 6062 thread = checkPlaybackThread_l(io); 6063 if (thread == NULL) { 6064 ALOGE("createEffect() unknown output thread"); 6065 lStatus = BAD_VALUE; 6066 goto Exit; 6067 } 6068 } 6069 6070 sp<Client> client = registerPid_l(pid); 6071 6072 // create effect on selected output thread 6073 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6074 &desc, enabled, &lStatus); 6075 if (handle != 0 && id != NULL) { 6076 *id = handle->id(); 6077 } 6078 } 6079 6080Exit: 6081 if(status) { 6082 *status = lStatus; 6083 } 6084 return handle; 6085} 6086 6087status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6088 audio_io_handle_t dstOutput) 6089{ 6090 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6091 sessionId, srcOutput, dstOutput); 6092 Mutex::Autolock _l(mLock); 6093 if (srcOutput == dstOutput) { 6094 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6095 return NO_ERROR; 6096 } 6097 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6098 if (srcThread == NULL) { 6099 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6100 return BAD_VALUE; 6101 } 6102 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6103 if (dstThread == NULL) { 6104 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6105 return BAD_VALUE; 6106 } 6107 6108 Mutex::Autolock _dl(dstThread->mLock); 6109 Mutex::Autolock _sl(srcThread->mLock); 6110 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6111 6112 return NO_ERROR; 6113} 6114 6115// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6116status_t AudioFlinger::moveEffectChain_l(int sessionId, 6117 AudioFlinger::PlaybackThread *srcThread, 6118 AudioFlinger::PlaybackThread *dstThread, 6119 bool reRegister) 6120{ 6121 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6122 sessionId, srcThread, dstThread); 6123 6124 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6125 if (chain == 0) { 6126 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6127 sessionId, srcThread); 6128 return INVALID_OPERATION; 6129 } 6130 6131 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6132 // so that a new chain is created with correct parameters when first effect is added. This is 6133 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6134 // removed. 6135 srcThread->removeEffectChain_l(chain); 6136 6137 // transfer all effects one by one so that new effect chain is created on new thread with 6138 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6139 audio_io_handle_t dstOutput = dstThread->id(); 6140 sp<EffectChain> dstChain; 6141 uint32_t strategy = 0; // prevent compiler warning 6142 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6143 while (effect != 0) { 6144 srcThread->removeEffect_l(effect); 6145 dstThread->addEffect_l(effect); 6146 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6147 if (effect->state() == EffectModule::ACTIVE || 6148 effect->state() == EffectModule::STOPPING) { 6149 effect->start(); 6150 } 6151 // if the move request is not received from audio policy manager, the effect must be 6152 // re-registered with the new strategy and output 6153 if (dstChain == 0) { 6154 dstChain = effect->chain().promote(); 6155 if (dstChain == 0) { 6156 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6157 srcThread->addEffect_l(effect); 6158 return NO_INIT; 6159 } 6160 strategy = dstChain->strategy(); 6161 } 6162 if (reRegister) { 6163 AudioSystem::unregisterEffect(effect->id()); 6164 AudioSystem::registerEffect(&effect->desc(), 6165 dstOutput, 6166 strategy, 6167 sessionId, 6168 effect->id()); 6169 } 6170 effect = chain->getEffectFromId_l(0); 6171 } 6172 6173 return NO_ERROR; 6174} 6175 6176 6177// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6178sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6179 const sp<AudioFlinger::Client>& client, 6180 const sp<IEffectClient>& effectClient, 6181 int32_t priority, 6182 int sessionId, 6183 effect_descriptor_t *desc, 6184 int *enabled, 6185 status_t *status 6186 ) 6187{ 6188 sp<EffectModule> effect; 6189 sp<EffectHandle> handle; 6190 status_t lStatus; 6191 sp<EffectChain> chain; 6192 bool chainCreated = false; 6193 bool effectCreated = false; 6194 bool effectRegistered = false; 6195 6196 lStatus = initCheck(); 6197 if (lStatus != NO_ERROR) { 6198 ALOGW("createEffect_l() Audio driver not initialized."); 6199 goto Exit; 6200 } 6201 6202 // Do not allow effects with session ID 0 on direct output or duplicating threads 6203 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6204 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6205 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6206 desc->name, sessionId); 6207 lStatus = BAD_VALUE; 6208 goto Exit; 6209 } 6210 // Only Pre processor effects are allowed on input threads and only on input threads 6211 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6212 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6213 desc->name, desc->flags, mType); 6214 lStatus = BAD_VALUE; 6215 goto Exit; 6216 } 6217 6218 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6219 6220 { // scope for mLock 6221 Mutex::Autolock _l(mLock); 6222 6223 // check for existing effect chain with the requested audio session 6224 chain = getEffectChain_l(sessionId); 6225 if (chain == 0) { 6226 // create a new chain for this session 6227 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6228 chain = new EffectChain(this, sessionId); 6229 addEffectChain_l(chain); 6230 chain->setStrategy(getStrategyForSession_l(sessionId)); 6231 chainCreated = true; 6232 } else { 6233 effect = chain->getEffectFromDesc_l(desc); 6234 } 6235 6236 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6237 6238 if (effect == 0) { 6239 int id = mAudioFlinger->nextUniqueId(); 6240 // Check CPU and memory usage 6241 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6242 if (lStatus != NO_ERROR) { 6243 goto Exit; 6244 } 6245 effectRegistered = true; 6246 // create a new effect module if none present in the chain 6247 effect = new EffectModule(this, chain, desc, id, sessionId); 6248 lStatus = effect->status(); 6249 if (lStatus != NO_ERROR) { 6250 goto Exit; 6251 } 6252 lStatus = chain->addEffect_l(effect); 6253 if (lStatus != NO_ERROR) { 6254 goto Exit; 6255 } 6256 effectCreated = true; 6257 6258 effect->setDevice(mDevice); 6259 effect->setMode(mAudioFlinger->getMode()); 6260 } 6261 // create effect handle and connect it to effect module 6262 handle = new EffectHandle(effect, client, effectClient, priority); 6263 lStatus = effect->addHandle(handle); 6264 if (enabled != NULL) { 6265 *enabled = (int)effect->isEnabled(); 6266 } 6267 } 6268 6269Exit: 6270 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6271 Mutex::Autolock _l(mLock); 6272 if (effectCreated) { 6273 chain->removeEffect_l(effect); 6274 } 6275 if (effectRegistered) { 6276 AudioSystem::unregisterEffect(effect->id()); 6277 } 6278 if (chainCreated) { 6279 removeEffectChain_l(chain); 6280 } 6281 handle.clear(); 6282 } 6283 6284 if(status) { 6285 *status = lStatus; 6286 } 6287 return handle; 6288} 6289 6290sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6291{ 6292 sp<EffectChain> chain = getEffectChain_l(sessionId); 6293 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6294} 6295 6296// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6297// PlaybackThread::mLock held 6298status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6299{ 6300 // check for existing effect chain with the requested audio session 6301 int sessionId = effect->sessionId(); 6302 sp<EffectChain> chain = getEffectChain_l(sessionId); 6303 bool chainCreated = false; 6304 6305 if (chain == 0) { 6306 // create a new chain for this session 6307 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6308 chain = new EffectChain(this, sessionId); 6309 addEffectChain_l(chain); 6310 chain->setStrategy(getStrategyForSession_l(sessionId)); 6311 chainCreated = true; 6312 } 6313 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6314 6315 if (chain->getEffectFromId_l(effect->id()) != 0) { 6316 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6317 this, effect->desc().name, chain.get()); 6318 return BAD_VALUE; 6319 } 6320 6321 status_t status = chain->addEffect_l(effect); 6322 if (status != NO_ERROR) { 6323 if (chainCreated) { 6324 removeEffectChain_l(chain); 6325 } 6326 return status; 6327 } 6328 6329 effect->setDevice(mDevice); 6330 effect->setMode(mAudioFlinger->getMode()); 6331 return NO_ERROR; 6332} 6333 6334void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6335 6336 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6337 effect_descriptor_t desc = effect->desc(); 6338 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6339 detachAuxEffect_l(effect->id()); 6340 } 6341 6342 sp<EffectChain> chain = effect->chain().promote(); 6343 if (chain != 0) { 6344 // remove effect chain if removing last effect 6345 if (chain->removeEffect_l(effect) == 0) { 6346 removeEffectChain_l(chain); 6347 } 6348 } else { 6349 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6350 } 6351} 6352 6353void AudioFlinger::ThreadBase::lockEffectChains_l( 6354 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6355{ 6356 effectChains = mEffectChains; 6357 for (size_t i = 0; i < mEffectChains.size(); i++) { 6358 mEffectChains[i]->lock(); 6359 } 6360} 6361 6362void AudioFlinger::ThreadBase::unlockEffectChains( 6363 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6364{ 6365 for (size_t i = 0; i < effectChains.size(); i++) { 6366 effectChains[i]->unlock(); 6367 } 6368} 6369 6370sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6371{ 6372 Mutex::Autolock _l(mLock); 6373 return getEffectChain_l(sessionId); 6374} 6375 6376sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6377{ 6378 size_t size = mEffectChains.size(); 6379 for (size_t i = 0; i < size; i++) { 6380 if (mEffectChains[i]->sessionId() == sessionId) { 6381 return mEffectChains[i]; 6382 } 6383 } 6384 return 0; 6385} 6386 6387void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6388{ 6389 Mutex::Autolock _l(mLock); 6390 size_t size = mEffectChains.size(); 6391 for (size_t i = 0; i < size; i++) { 6392 mEffectChains[i]->setMode_l(mode); 6393 } 6394} 6395 6396void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6397 const wp<EffectHandle>& handle, 6398 bool unpinIfLast) { 6399 6400 Mutex::Autolock _l(mLock); 6401 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6402 // delete the effect module if removing last handle on it 6403 if (effect->removeHandle(handle) == 0) { 6404 if (!effect->isPinned() || unpinIfLast) { 6405 removeEffect_l(effect); 6406 AudioSystem::unregisterEffect(effect->id()); 6407 } 6408 } 6409} 6410 6411status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6412{ 6413 int session = chain->sessionId(); 6414 int16_t *buffer = mMixBuffer; 6415 bool ownsBuffer = false; 6416 6417 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6418 if (session > 0) { 6419 // Only one effect chain can be present in direct output thread and it uses 6420 // the mix buffer as input 6421 if (mType != DIRECT) { 6422 size_t numSamples = mFrameCount * mChannelCount; 6423 buffer = new int16_t[numSamples]; 6424 memset(buffer, 0, numSamples * sizeof(int16_t)); 6425 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6426 ownsBuffer = true; 6427 } 6428 6429 // Attach all tracks with same session ID to this chain. 6430 for (size_t i = 0; i < mTracks.size(); ++i) { 6431 sp<Track> track = mTracks[i]; 6432 if (session == track->sessionId()) { 6433 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6434 track->setMainBuffer(buffer); 6435 chain->incTrackCnt(); 6436 } 6437 } 6438 6439 // indicate all active tracks in the chain 6440 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6441 sp<Track> track = mActiveTracks[i].promote(); 6442 if (track == 0) continue; 6443 if (session == track->sessionId()) { 6444 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6445 chain->incActiveTrackCnt(); 6446 } 6447 } 6448 } 6449 6450 chain->setInBuffer(buffer, ownsBuffer); 6451 chain->setOutBuffer(mMixBuffer); 6452 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6453 // chains list in order to be processed last as it contains output stage effects 6454 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6455 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6456 // after track specific effects and before output stage 6457 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6458 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6459 // Effect chain for other sessions are inserted at beginning of effect 6460 // chains list to be processed before output mix effects. Relative order between other 6461 // sessions is not important 6462 size_t size = mEffectChains.size(); 6463 size_t i = 0; 6464 for (i = 0; i < size; i++) { 6465 if (mEffectChains[i]->sessionId() < session) break; 6466 } 6467 mEffectChains.insertAt(chain, i); 6468 checkSuspendOnAddEffectChain_l(chain); 6469 6470 return NO_ERROR; 6471} 6472 6473size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6474{ 6475 int session = chain->sessionId(); 6476 6477 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6478 6479 for (size_t i = 0; i < mEffectChains.size(); i++) { 6480 if (chain == mEffectChains[i]) { 6481 mEffectChains.removeAt(i); 6482 // detach all active tracks from the chain 6483 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6484 sp<Track> track = mActiveTracks[i].promote(); 6485 if (track == 0) continue; 6486 if (session == track->sessionId()) { 6487 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6488 chain.get(), session); 6489 chain->decActiveTrackCnt(); 6490 } 6491 } 6492 6493 // detach all tracks with same session ID from this chain 6494 for (size_t i = 0; i < mTracks.size(); ++i) { 6495 sp<Track> track = mTracks[i]; 6496 if (session == track->sessionId()) { 6497 track->setMainBuffer(mMixBuffer); 6498 chain->decTrackCnt(); 6499 } 6500 } 6501 break; 6502 } 6503 } 6504 return mEffectChains.size(); 6505} 6506 6507status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6508 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6509{ 6510 Mutex::Autolock _l(mLock); 6511 return attachAuxEffect_l(track, EffectId); 6512} 6513 6514status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6515 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6516{ 6517 status_t status = NO_ERROR; 6518 6519 if (EffectId == 0) { 6520 track->setAuxBuffer(0, NULL); 6521 } else { 6522 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6523 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6524 if (effect != 0) { 6525 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6526 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6527 } else { 6528 status = INVALID_OPERATION; 6529 } 6530 } else { 6531 status = BAD_VALUE; 6532 } 6533 } 6534 return status; 6535} 6536 6537void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6538{ 6539 for (size_t i = 0; i < mTracks.size(); ++i) { 6540 sp<Track> track = mTracks[i]; 6541 if (track->auxEffectId() == effectId) { 6542 attachAuxEffect_l(track, 0); 6543 } 6544 } 6545} 6546 6547status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6548{ 6549 // only one chain per input thread 6550 if (mEffectChains.size() != 0) { 6551 return INVALID_OPERATION; 6552 } 6553 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6554 6555 chain->setInBuffer(NULL); 6556 chain->setOutBuffer(NULL); 6557 6558 checkSuspendOnAddEffectChain_l(chain); 6559 6560 mEffectChains.add(chain); 6561 6562 return NO_ERROR; 6563} 6564 6565size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6566{ 6567 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6568 ALOGW_IF(mEffectChains.size() != 1, 6569 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6570 chain.get(), mEffectChains.size(), this); 6571 if (mEffectChains.size() == 1) { 6572 mEffectChains.removeAt(0); 6573 } 6574 return 0; 6575} 6576 6577// ---------------------------------------------------------------------------- 6578// EffectModule implementation 6579// ---------------------------------------------------------------------------- 6580 6581#undef LOG_TAG 6582#define LOG_TAG "AudioFlinger::EffectModule" 6583 6584AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6585 const wp<AudioFlinger::EffectChain>& chain, 6586 effect_descriptor_t *desc, 6587 int id, 6588 int sessionId) 6589 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6590 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6591{ 6592 ALOGV("Constructor %p", this); 6593 int lStatus; 6594 if (thread == NULL) { 6595 return; 6596 } 6597 6598 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6599 6600 // create effect engine from effect factory 6601 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6602 6603 if (mStatus != NO_ERROR) { 6604 return; 6605 } 6606 lStatus = init(); 6607 if (lStatus < 0) { 6608 mStatus = lStatus; 6609 goto Error; 6610 } 6611 6612 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6613 mPinned = true; 6614 } 6615 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6616 return; 6617Error: 6618 EffectRelease(mEffectInterface); 6619 mEffectInterface = NULL; 6620 ALOGV("Constructor Error %d", mStatus); 6621} 6622 6623AudioFlinger::EffectModule::~EffectModule() 6624{ 6625 ALOGV("Destructor %p", this); 6626 if (mEffectInterface != NULL) { 6627 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6628 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6629 sp<ThreadBase> thread = mThread.promote(); 6630 if (thread != 0) { 6631 audio_stream_t *stream = thread->stream(); 6632 if (stream != NULL) { 6633 stream->remove_audio_effect(stream, mEffectInterface); 6634 } 6635 } 6636 } 6637 // release effect engine 6638 EffectRelease(mEffectInterface); 6639 } 6640} 6641 6642status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6643{ 6644 status_t status; 6645 6646 Mutex::Autolock _l(mLock); 6647 int priority = handle->priority(); 6648 size_t size = mHandles.size(); 6649 sp<EffectHandle> h; 6650 size_t i; 6651 for (i = 0; i < size; i++) { 6652 h = mHandles[i].promote(); 6653 if (h == 0) continue; 6654 if (h->priority() <= priority) break; 6655 } 6656 // if inserted in first place, move effect control from previous owner to this handle 6657 if (i == 0) { 6658 bool enabled = false; 6659 if (h != 0) { 6660 enabled = h->enabled(); 6661 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6662 } 6663 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6664 status = NO_ERROR; 6665 } else { 6666 status = ALREADY_EXISTS; 6667 } 6668 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6669 mHandles.insertAt(handle, i); 6670 return status; 6671} 6672 6673size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6674{ 6675 Mutex::Autolock _l(mLock); 6676 size_t size = mHandles.size(); 6677 size_t i; 6678 for (i = 0; i < size; i++) { 6679 if (mHandles[i] == handle) break; 6680 } 6681 if (i == size) { 6682 return size; 6683 } 6684 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6685 6686 bool enabled = false; 6687 EffectHandle *hdl = handle.unsafe_get(); 6688 if (hdl != NULL) { 6689 ALOGV("removeHandle() unsafe_get OK"); 6690 enabled = hdl->enabled(); 6691 } 6692 mHandles.removeAt(i); 6693 size = mHandles.size(); 6694 // if removed from first place, move effect control from this handle to next in line 6695 if (i == 0 && size != 0) { 6696 sp<EffectHandle> h = mHandles[0].promote(); 6697 if (h != 0) { 6698 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6699 } 6700 } 6701 6702 // Prevent calls to process() and other functions on effect interface from now on. 6703 // The effect engine will be released by the destructor when the last strong reference on 6704 // this object is released which can happen after next process is called. 6705 if (size == 0 && !mPinned) { 6706 mState = DESTROYED; 6707 } 6708 6709 return size; 6710} 6711 6712sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6713{ 6714 Mutex::Autolock _l(mLock); 6715 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6716} 6717 6718void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6719{ 6720 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6721 // keep a strong reference on this EffectModule to avoid calling the 6722 // destructor before we exit 6723 sp<EffectModule> keep(this); 6724 { 6725 sp<ThreadBase> thread = mThread.promote(); 6726 if (thread != 0) { 6727 thread->disconnectEffect(keep, handle, unpinIfLast); 6728 } 6729 } 6730} 6731 6732void AudioFlinger::EffectModule::updateState() { 6733 Mutex::Autolock _l(mLock); 6734 6735 switch (mState) { 6736 case RESTART: 6737 reset_l(); 6738 // FALL THROUGH 6739 6740 case STARTING: 6741 // clear auxiliary effect input buffer for next accumulation 6742 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6743 memset(mConfig.inputCfg.buffer.raw, 6744 0, 6745 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6746 } 6747 start_l(); 6748 mState = ACTIVE; 6749 break; 6750 case STOPPING: 6751 stop_l(); 6752 mDisableWaitCnt = mMaxDisableWaitCnt; 6753 mState = STOPPED; 6754 break; 6755 case STOPPED: 6756 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6757 // turn off sequence. 6758 if (--mDisableWaitCnt == 0) { 6759 reset_l(); 6760 mState = IDLE; 6761 } 6762 break; 6763 default: //IDLE , ACTIVE, DESTROYED 6764 break; 6765 } 6766} 6767 6768void AudioFlinger::EffectModule::process() 6769{ 6770 Mutex::Autolock _l(mLock); 6771 6772 if (mState == DESTROYED || mEffectInterface == NULL || 6773 mConfig.inputCfg.buffer.raw == NULL || 6774 mConfig.outputCfg.buffer.raw == NULL) { 6775 return; 6776 } 6777 6778 if (isProcessEnabled()) { 6779 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6780 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6781 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6782 mConfig.inputCfg.buffer.s32, 6783 mConfig.inputCfg.buffer.frameCount/2); 6784 } 6785 6786 // do the actual processing in the effect engine 6787 int ret = (*mEffectInterface)->process(mEffectInterface, 6788 &mConfig.inputCfg.buffer, 6789 &mConfig.outputCfg.buffer); 6790 6791 // force transition to IDLE state when engine is ready 6792 if (mState == STOPPED && ret == -ENODATA) { 6793 mDisableWaitCnt = 1; 6794 } 6795 6796 // clear auxiliary effect input buffer for next accumulation 6797 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6798 memset(mConfig.inputCfg.buffer.raw, 0, 6799 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6800 } 6801 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6802 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6803 // If an insert effect is idle and input buffer is different from output buffer, 6804 // accumulate input onto output 6805 sp<EffectChain> chain = mChain.promote(); 6806 if (chain != 0 && chain->activeTrackCnt() != 0) { 6807 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6808 int16_t *in = mConfig.inputCfg.buffer.s16; 6809 int16_t *out = mConfig.outputCfg.buffer.s16; 6810 for (size_t i = 0; i < frameCnt; i++) { 6811 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6812 } 6813 } 6814 } 6815} 6816 6817void AudioFlinger::EffectModule::reset_l() 6818{ 6819 if (mEffectInterface == NULL) { 6820 return; 6821 } 6822 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6823} 6824 6825status_t AudioFlinger::EffectModule::configure() 6826{ 6827 uint32_t channels; 6828 if (mEffectInterface == NULL) { 6829 return NO_INIT; 6830 } 6831 6832 sp<ThreadBase> thread = mThread.promote(); 6833 if (thread == 0) { 6834 return DEAD_OBJECT; 6835 } 6836 6837 // TODO: handle configuration of effects replacing track process 6838 if (thread->channelCount() == 1) { 6839 channels = AUDIO_CHANNEL_OUT_MONO; 6840 } else { 6841 channels = AUDIO_CHANNEL_OUT_STEREO; 6842 } 6843 6844 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6845 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6846 } else { 6847 mConfig.inputCfg.channels = channels; 6848 } 6849 mConfig.outputCfg.channels = channels; 6850 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6851 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6852 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6853 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6854 mConfig.inputCfg.bufferProvider.cookie = NULL; 6855 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6856 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6857 mConfig.outputCfg.bufferProvider.cookie = NULL; 6858 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6859 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6860 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6861 // Insert effect: 6862 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6863 // always overwrites output buffer: input buffer == output buffer 6864 // - in other sessions: 6865 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6866 // other effect: overwrites output buffer: input buffer == output buffer 6867 // Auxiliary effect: 6868 // accumulates in output buffer: input buffer != output buffer 6869 // Therefore: accumulate <=> input buffer != output buffer 6870 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6871 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6872 } else { 6873 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6874 } 6875 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6876 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6877 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6878 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6879 6880 ALOGV("configure() %p thread %p buffer %p framecount %d", 6881 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6882 6883 status_t cmdStatus; 6884 uint32_t size = sizeof(int); 6885 status_t status = (*mEffectInterface)->command(mEffectInterface, 6886 EFFECT_CMD_SET_CONFIG, 6887 sizeof(effect_config_t), 6888 &mConfig, 6889 &size, 6890 &cmdStatus); 6891 if (status == 0) { 6892 status = cmdStatus; 6893 } 6894 6895 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6896 (1000 * mConfig.outputCfg.buffer.frameCount); 6897 6898 return status; 6899} 6900 6901status_t AudioFlinger::EffectModule::init() 6902{ 6903 Mutex::Autolock _l(mLock); 6904 if (mEffectInterface == NULL) { 6905 return NO_INIT; 6906 } 6907 status_t cmdStatus; 6908 uint32_t size = sizeof(status_t); 6909 status_t status = (*mEffectInterface)->command(mEffectInterface, 6910 EFFECT_CMD_INIT, 6911 0, 6912 NULL, 6913 &size, 6914 &cmdStatus); 6915 if (status == 0) { 6916 status = cmdStatus; 6917 } 6918 return status; 6919} 6920 6921status_t AudioFlinger::EffectModule::start() 6922{ 6923 Mutex::Autolock _l(mLock); 6924 return start_l(); 6925} 6926 6927status_t AudioFlinger::EffectModule::start_l() 6928{ 6929 if (mEffectInterface == NULL) { 6930 return NO_INIT; 6931 } 6932 status_t cmdStatus; 6933 uint32_t size = sizeof(status_t); 6934 status_t status = (*mEffectInterface)->command(mEffectInterface, 6935 EFFECT_CMD_ENABLE, 6936 0, 6937 NULL, 6938 &size, 6939 &cmdStatus); 6940 if (status == 0) { 6941 status = cmdStatus; 6942 } 6943 if (status == 0 && 6944 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6945 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6946 sp<ThreadBase> thread = mThread.promote(); 6947 if (thread != 0) { 6948 audio_stream_t *stream = thread->stream(); 6949 if (stream != NULL) { 6950 stream->add_audio_effect(stream, mEffectInterface); 6951 } 6952 } 6953 } 6954 return status; 6955} 6956 6957status_t AudioFlinger::EffectModule::stop() 6958{ 6959 Mutex::Autolock _l(mLock); 6960 return stop_l(); 6961} 6962 6963status_t AudioFlinger::EffectModule::stop_l() 6964{ 6965 if (mEffectInterface == NULL) { 6966 return NO_INIT; 6967 } 6968 status_t cmdStatus; 6969 uint32_t size = sizeof(status_t); 6970 status_t status = (*mEffectInterface)->command(mEffectInterface, 6971 EFFECT_CMD_DISABLE, 6972 0, 6973 NULL, 6974 &size, 6975 &cmdStatus); 6976 if (status == 0) { 6977 status = cmdStatus; 6978 } 6979 if (status == 0 && 6980 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6981 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6982 sp<ThreadBase> thread = mThread.promote(); 6983 if (thread != 0) { 6984 audio_stream_t *stream = thread->stream(); 6985 if (stream != NULL) { 6986 stream->remove_audio_effect(stream, mEffectInterface); 6987 } 6988 } 6989 } 6990 return status; 6991} 6992 6993status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6994 uint32_t cmdSize, 6995 void *pCmdData, 6996 uint32_t *replySize, 6997 void *pReplyData) 6998{ 6999 Mutex::Autolock _l(mLock); 7000// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7001 7002 if (mState == DESTROYED || mEffectInterface == NULL) { 7003 return NO_INIT; 7004 } 7005 status_t status = (*mEffectInterface)->command(mEffectInterface, 7006 cmdCode, 7007 cmdSize, 7008 pCmdData, 7009 replySize, 7010 pReplyData); 7011 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7012 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7013 for (size_t i = 1; i < mHandles.size(); i++) { 7014 sp<EffectHandle> h = mHandles[i].promote(); 7015 if (h != 0) { 7016 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7017 } 7018 } 7019 } 7020 return status; 7021} 7022 7023status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7024{ 7025 7026 Mutex::Autolock _l(mLock); 7027 ALOGV("setEnabled %p enabled %d", this, enabled); 7028 7029 if (enabled != isEnabled()) { 7030 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7031 if (enabled && status != NO_ERROR) { 7032 return status; 7033 } 7034 7035 switch (mState) { 7036 // going from disabled to enabled 7037 case IDLE: 7038 mState = STARTING; 7039 break; 7040 case STOPPED: 7041 mState = RESTART; 7042 break; 7043 case STOPPING: 7044 mState = ACTIVE; 7045 break; 7046 7047 // going from enabled to disabled 7048 case RESTART: 7049 mState = STOPPED; 7050 break; 7051 case STARTING: 7052 mState = IDLE; 7053 break; 7054 case ACTIVE: 7055 mState = STOPPING; 7056 break; 7057 case DESTROYED: 7058 return NO_ERROR; // simply ignore as we are being destroyed 7059 } 7060 for (size_t i = 1; i < mHandles.size(); i++) { 7061 sp<EffectHandle> h = mHandles[i].promote(); 7062 if (h != 0) { 7063 h->setEnabled(enabled); 7064 } 7065 } 7066 } 7067 return NO_ERROR; 7068} 7069 7070bool AudioFlinger::EffectModule::isEnabled() const 7071{ 7072 switch (mState) { 7073 case RESTART: 7074 case STARTING: 7075 case ACTIVE: 7076 return true; 7077 case IDLE: 7078 case STOPPING: 7079 case STOPPED: 7080 case DESTROYED: 7081 default: 7082 return false; 7083 } 7084} 7085 7086bool AudioFlinger::EffectModule::isProcessEnabled() const 7087{ 7088 switch (mState) { 7089 case RESTART: 7090 case ACTIVE: 7091 case STOPPING: 7092 case STOPPED: 7093 return true; 7094 case IDLE: 7095 case STARTING: 7096 case DESTROYED: 7097 default: 7098 return false; 7099 } 7100} 7101 7102status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7103{ 7104 Mutex::Autolock _l(mLock); 7105 status_t status = NO_ERROR; 7106 7107 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7108 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7109 if (isProcessEnabled() && 7110 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7111 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7112 status_t cmdStatus; 7113 uint32_t volume[2]; 7114 uint32_t *pVolume = NULL; 7115 uint32_t size = sizeof(volume); 7116 volume[0] = *left; 7117 volume[1] = *right; 7118 if (controller) { 7119 pVolume = volume; 7120 } 7121 status = (*mEffectInterface)->command(mEffectInterface, 7122 EFFECT_CMD_SET_VOLUME, 7123 size, 7124 volume, 7125 &size, 7126 pVolume); 7127 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7128 *left = volume[0]; 7129 *right = volume[1]; 7130 } 7131 } 7132 return status; 7133} 7134 7135status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7136{ 7137 Mutex::Autolock _l(mLock); 7138 status_t status = NO_ERROR; 7139 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7140 // audio pre processing modules on RecordThread can receive both output and 7141 // input device indication in the same call 7142 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7143 if (dev) { 7144 status_t cmdStatus; 7145 uint32_t size = sizeof(status_t); 7146 7147 status = (*mEffectInterface)->command(mEffectInterface, 7148 EFFECT_CMD_SET_DEVICE, 7149 sizeof(uint32_t), 7150 &dev, 7151 &size, 7152 &cmdStatus); 7153 if (status == NO_ERROR) { 7154 status = cmdStatus; 7155 } 7156 } 7157 dev = device & AUDIO_DEVICE_IN_ALL; 7158 if (dev) { 7159 status_t cmdStatus; 7160 uint32_t size = sizeof(status_t); 7161 7162 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7163 EFFECT_CMD_SET_INPUT_DEVICE, 7164 sizeof(uint32_t), 7165 &dev, 7166 &size, 7167 &cmdStatus); 7168 if (status2 == NO_ERROR) { 7169 status2 = cmdStatus; 7170 } 7171 if (status == NO_ERROR) { 7172 status = status2; 7173 } 7174 } 7175 } 7176 return status; 7177} 7178 7179status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7180{ 7181 Mutex::Autolock _l(mLock); 7182 status_t status = NO_ERROR; 7183 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7184 status_t cmdStatus; 7185 uint32_t size = sizeof(status_t); 7186 status = (*mEffectInterface)->command(mEffectInterface, 7187 EFFECT_CMD_SET_AUDIO_MODE, 7188 sizeof(audio_mode_t), 7189 &mode, 7190 &size, 7191 &cmdStatus); 7192 if (status == NO_ERROR) { 7193 status = cmdStatus; 7194 } 7195 } 7196 return status; 7197} 7198 7199void AudioFlinger::EffectModule::setSuspended(bool suspended) 7200{ 7201 Mutex::Autolock _l(mLock); 7202 mSuspended = suspended; 7203} 7204 7205bool AudioFlinger::EffectModule::suspended() const 7206{ 7207 Mutex::Autolock _l(mLock); 7208 return mSuspended; 7209} 7210 7211status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7212{ 7213 const size_t SIZE = 256; 7214 char buffer[SIZE]; 7215 String8 result; 7216 7217 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7218 result.append(buffer); 7219 7220 bool locked = tryLock(mLock); 7221 // failed to lock - AudioFlinger is probably deadlocked 7222 if (!locked) { 7223 result.append("\t\tCould not lock Fx mutex:\n"); 7224 } 7225 7226 result.append("\t\tSession Status State Engine:\n"); 7227 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7228 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7229 result.append(buffer); 7230 7231 result.append("\t\tDescriptor:\n"); 7232 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7233 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7234 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7235 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7236 result.append(buffer); 7237 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7238 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7239 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7240 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7241 result.append(buffer); 7242 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7243 mDescriptor.apiVersion, 7244 mDescriptor.flags); 7245 result.append(buffer); 7246 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7247 mDescriptor.name); 7248 result.append(buffer); 7249 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7250 mDescriptor.implementor); 7251 result.append(buffer); 7252 7253 result.append("\t\t- Input configuration:\n"); 7254 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7255 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7256 (uint32_t)mConfig.inputCfg.buffer.raw, 7257 mConfig.inputCfg.buffer.frameCount, 7258 mConfig.inputCfg.samplingRate, 7259 mConfig.inputCfg.channels, 7260 mConfig.inputCfg.format); 7261 result.append(buffer); 7262 7263 result.append("\t\t- Output configuration:\n"); 7264 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7265 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7266 (uint32_t)mConfig.outputCfg.buffer.raw, 7267 mConfig.outputCfg.buffer.frameCount, 7268 mConfig.outputCfg.samplingRate, 7269 mConfig.outputCfg.channels, 7270 mConfig.outputCfg.format); 7271 result.append(buffer); 7272 7273 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7274 result.append(buffer); 7275 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7276 for (size_t i = 0; i < mHandles.size(); ++i) { 7277 sp<EffectHandle> handle = mHandles[i].promote(); 7278 if (handle != 0) { 7279 handle->dump(buffer, SIZE); 7280 result.append(buffer); 7281 } 7282 } 7283 7284 result.append("\n"); 7285 7286 write(fd, result.string(), result.length()); 7287 7288 if (locked) { 7289 mLock.unlock(); 7290 } 7291 7292 return NO_ERROR; 7293} 7294 7295// ---------------------------------------------------------------------------- 7296// EffectHandle implementation 7297// ---------------------------------------------------------------------------- 7298 7299#undef LOG_TAG 7300#define LOG_TAG "AudioFlinger::EffectHandle" 7301 7302AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7303 const sp<AudioFlinger::Client>& client, 7304 const sp<IEffectClient>& effectClient, 7305 int32_t priority) 7306 : BnEffect(), 7307 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7308 mPriority(priority), mHasControl(false), mEnabled(false) 7309{ 7310 ALOGV("constructor %p", this); 7311 7312 if (client == 0) { 7313 return; 7314 } 7315 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7316 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7317 if (mCblkMemory != 0) { 7318 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7319 7320 if (mCblk != NULL) { 7321 new(mCblk) effect_param_cblk_t(); 7322 mBuffer = (uint8_t *)mCblk + bufOffset; 7323 } 7324 } else { 7325 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7326 return; 7327 } 7328} 7329 7330AudioFlinger::EffectHandle::~EffectHandle() 7331{ 7332 ALOGV("Destructor %p", this); 7333 disconnect(false); 7334 ALOGV("Destructor DONE %p", this); 7335} 7336 7337status_t AudioFlinger::EffectHandle::enable() 7338{ 7339 ALOGV("enable %p", this); 7340 if (!mHasControl) return INVALID_OPERATION; 7341 if (mEffect == 0) return DEAD_OBJECT; 7342 7343 if (mEnabled) { 7344 return NO_ERROR; 7345 } 7346 7347 mEnabled = true; 7348 7349 sp<ThreadBase> thread = mEffect->thread().promote(); 7350 if (thread != 0) { 7351 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7352 } 7353 7354 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7355 if (mEffect->suspended()) { 7356 return NO_ERROR; 7357 } 7358 7359 status_t status = mEffect->setEnabled(true); 7360 if (status != NO_ERROR) { 7361 if (thread != 0) { 7362 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7363 } 7364 mEnabled = false; 7365 } 7366 return status; 7367} 7368 7369status_t AudioFlinger::EffectHandle::disable() 7370{ 7371 ALOGV("disable %p", this); 7372 if (!mHasControl) return INVALID_OPERATION; 7373 if (mEffect == 0) return DEAD_OBJECT; 7374 7375 if (!mEnabled) { 7376 return NO_ERROR; 7377 } 7378 mEnabled = false; 7379 7380 if (mEffect->suspended()) { 7381 return NO_ERROR; 7382 } 7383 7384 status_t status = mEffect->setEnabled(false); 7385 7386 sp<ThreadBase> thread = mEffect->thread().promote(); 7387 if (thread != 0) { 7388 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7389 } 7390 7391 return status; 7392} 7393 7394void AudioFlinger::EffectHandle::disconnect() 7395{ 7396 disconnect(true); 7397} 7398 7399void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7400{ 7401 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7402 if (mEffect == 0) { 7403 return; 7404 } 7405 mEffect->disconnect(this, unpinIfLast); 7406 7407 if (mHasControl && mEnabled) { 7408 sp<ThreadBase> thread = mEffect->thread().promote(); 7409 if (thread != 0) { 7410 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7411 } 7412 } 7413 7414 // release sp on module => module destructor can be called now 7415 mEffect.clear(); 7416 if (mClient != 0) { 7417 if (mCblk != NULL) { 7418 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7419 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7420 } 7421 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7422 // Client destructor must run with AudioFlinger mutex locked 7423 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7424 mClient.clear(); 7425 } 7426} 7427 7428status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7429 uint32_t cmdSize, 7430 void *pCmdData, 7431 uint32_t *replySize, 7432 void *pReplyData) 7433{ 7434// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7435// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7436 7437 // only get parameter command is permitted for applications not controlling the effect 7438 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7439 return INVALID_OPERATION; 7440 } 7441 if (mEffect == 0) return DEAD_OBJECT; 7442 if (mClient == 0) return INVALID_OPERATION; 7443 7444 // handle commands that are not forwarded transparently to effect engine 7445 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7446 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7447 // no risk to block the whole media server process or mixer threads is we are stuck here 7448 Mutex::Autolock _l(mCblk->lock); 7449 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7450 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7451 mCblk->serverIndex = 0; 7452 mCblk->clientIndex = 0; 7453 return BAD_VALUE; 7454 } 7455 status_t status = NO_ERROR; 7456 while (mCblk->serverIndex < mCblk->clientIndex) { 7457 int reply; 7458 uint32_t rsize = sizeof(int); 7459 int *p = (int *)(mBuffer + mCblk->serverIndex); 7460 int size = *p++; 7461 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7462 ALOGW("command(): invalid parameter block size"); 7463 break; 7464 } 7465 effect_param_t *param = (effect_param_t *)p; 7466 if (param->psize == 0 || param->vsize == 0) { 7467 ALOGW("command(): null parameter or value size"); 7468 mCblk->serverIndex += size; 7469 continue; 7470 } 7471 uint32_t psize = sizeof(effect_param_t) + 7472 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7473 param->vsize; 7474 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7475 psize, 7476 p, 7477 &rsize, 7478 &reply); 7479 // stop at first error encountered 7480 if (ret != NO_ERROR) { 7481 status = ret; 7482 *(int *)pReplyData = reply; 7483 break; 7484 } else if (reply != NO_ERROR) { 7485 *(int *)pReplyData = reply; 7486 break; 7487 } 7488 mCblk->serverIndex += size; 7489 } 7490 mCblk->serverIndex = 0; 7491 mCblk->clientIndex = 0; 7492 return status; 7493 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7494 *(int *)pReplyData = NO_ERROR; 7495 return enable(); 7496 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7497 *(int *)pReplyData = NO_ERROR; 7498 return disable(); 7499 } 7500 7501 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7502} 7503 7504void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7505{ 7506 ALOGV("setControl %p control %d", this, hasControl); 7507 7508 mHasControl = hasControl; 7509 mEnabled = enabled; 7510 7511 if (signal && mEffectClient != 0) { 7512 mEffectClient->controlStatusChanged(hasControl); 7513 } 7514} 7515 7516void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7517 uint32_t cmdSize, 7518 void *pCmdData, 7519 uint32_t replySize, 7520 void *pReplyData) 7521{ 7522 if (mEffectClient != 0) { 7523 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7524 } 7525} 7526 7527 7528 7529void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7530{ 7531 if (mEffectClient != 0) { 7532 mEffectClient->enableStatusChanged(enabled); 7533 } 7534} 7535 7536status_t AudioFlinger::EffectHandle::onTransact( 7537 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7538{ 7539 return BnEffect::onTransact(code, data, reply, flags); 7540} 7541 7542 7543void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7544{ 7545 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7546 7547 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7548 (mClient == 0) ? getpid_cached : mClient->pid(), 7549 mPriority, 7550 mHasControl, 7551 !locked, 7552 mCblk ? mCblk->clientIndex : 0, 7553 mCblk ? mCblk->serverIndex : 0 7554 ); 7555 7556 if (locked) { 7557 mCblk->lock.unlock(); 7558 } 7559} 7560 7561#undef LOG_TAG 7562#define LOG_TAG "AudioFlinger::EffectChain" 7563 7564AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7565 int sessionId) 7566 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7567 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7568 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7569{ 7570 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7571 if (thread == NULL) { 7572 return; 7573 } 7574 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7575 thread->frameCount(); 7576} 7577 7578AudioFlinger::EffectChain::~EffectChain() 7579{ 7580 if (mOwnInBuffer) { 7581 delete mInBuffer; 7582 } 7583 7584} 7585 7586// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7587sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7588{ 7589 size_t size = mEffects.size(); 7590 7591 for (size_t i = 0; i < size; i++) { 7592 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7593 return mEffects[i]; 7594 } 7595 } 7596 return 0; 7597} 7598 7599// getEffectFromId_l() must be called with ThreadBase::mLock held 7600sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7601{ 7602 size_t size = mEffects.size(); 7603 7604 for (size_t i = 0; i < size; i++) { 7605 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7606 if (id == 0 || mEffects[i]->id() == id) { 7607 return mEffects[i]; 7608 } 7609 } 7610 return 0; 7611} 7612 7613// getEffectFromType_l() must be called with ThreadBase::mLock held 7614sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7615 const effect_uuid_t *type) 7616{ 7617 size_t size = mEffects.size(); 7618 7619 for (size_t i = 0; i < size; i++) { 7620 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7621 return mEffects[i]; 7622 } 7623 } 7624 return 0; 7625} 7626 7627// Must be called with EffectChain::mLock locked 7628void AudioFlinger::EffectChain::process_l() 7629{ 7630 sp<ThreadBase> thread = mThread.promote(); 7631 if (thread == 0) { 7632 ALOGW("process_l(): cannot promote mixer thread"); 7633 return; 7634 } 7635 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7636 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7637 // always process effects unless no more tracks are on the session and the effect tail 7638 // has been rendered 7639 bool doProcess = true; 7640 if (!isGlobalSession) { 7641 bool tracksOnSession = (trackCnt() != 0); 7642 7643 if (!tracksOnSession && mTailBufferCount == 0) { 7644 doProcess = false; 7645 } 7646 7647 if (activeTrackCnt() == 0) { 7648 // if no track is active and the effect tail has not been rendered, 7649 // the input buffer must be cleared here as the mixer process will not do it 7650 if (tracksOnSession || mTailBufferCount > 0) { 7651 size_t numSamples = thread->frameCount() * thread->channelCount(); 7652 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7653 if (mTailBufferCount > 0) { 7654 mTailBufferCount--; 7655 } 7656 } 7657 } 7658 } 7659 7660 size_t size = mEffects.size(); 7661 if (doProcess) { 7662 for (size_t i = 0; i < size; i++) { 7663 mEffects[i]->process(); 7664 } 7665 } 7666 for (size_t i = 0; i < size; i++) { 7667 mEffects[i]->updateState(); 7668 } 7669} 7670 7671// addEffect_l() must be called with PlaybackThread::mLock held 7672status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7673{ 7674 effect_descriptor_t desc = effect->desc(); 7675 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7676 7677 Mutex::Autolock _l(mLock); 7678 effect->setChain(this); 7679 sp<ThreadBase> thread = mThread.promote(); 7680 if (thread == 0) { 7681 return NO_INIT; 7682 } 7683 effect->setThread(thread); 7684 7685 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7686 // Auxiliary effects are inserted at the beginning of mEffects vector as 7687 // they are processed first and accumulated in chain input buffer 7688 mEffects.insertAt(effect, 0); 7689 7690 // the input buffer for auxiliary effect contains mono samples in 7691 // 32 bit format. This is to avoid saturation in AudoMixer 7692 // accumulation stage. Saturation is done in EffectModule::process() before 7693 // calling the process in effect engine 7694 size_t numSamples = thread->frameCount(); 7695 int32_t *buffer = new int32_t[numSamples]; 7696 memset(buffer, 0, numSamples * sizeof(int32_t)); 7697 effect->setInBuffer((int16_t *)buffer); 7698 // auxiliary effects output samples to chain input buffer for further processing 7699 // by insert effects 7700 effect->setOutBuffer(mInBuffer); 7701 } else { 7702 // Insert effects are inserted at the end of mEffects vector as they are processed 7703 // after track and auxiliary effects. 7704 // Insert effect order as a function of indicated preference: 7705 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7706 // another effect is present 7707 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7708 // last effect claiming first position 7709 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7710 // first effect claiming last position 7711 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7712 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7713 // already present 7714 7715 size_t size = mEffects.size(); 7716 size_t idx_insert = size; 7717 ssize_t idx_insert_first = -1; 7718 ssize_t idx_insert_last = -1; 7719 7720 for (size_t i = 0; i < size; i++) { 7721 effect_descriptor_t d = mEffects[i]->desc(); 7722 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7723 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7724 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7725 // check invalid effect chaining combinations 7726 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7727 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7728 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7729 return INVALID_OPERATION; 7730 } 7731 // remember position of first insert effect and by default 7732 // select this as insert position for new effect 7733 if (idx_insert == size) { 7734 idx_insert = i; 7735 } 7736 // remember position of last insert effect claiming 7737 // first position 7738 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7739 idx_insert_first = i; 7740 } 7741 // remember position of first insert effect claiming 7742 // last position 7743 if (iPref == EFFECT_FLAG_INSERT_LAST && 7744 idx_insert_last == -1) { 7745 idx_insert_last = i; 7746 } 7747 } 7748 } 7749 7750 // modify idx_insert from first position if needed 7751 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7752 if (idx_insert_last != -1) { 7753 idx_insert = idx_insert_last; 7754 } else { 7755 idx_insert = size; 7756 } 7757 } else { 7758 if (idx_insert_first != -1) { 7759 idx_insert = idx_insert_first + 1; 7760 } 7761 } 7762 7763 // always read samples from chain input buffer 7764 effect->setInBuffer(mInBuffer); 7765 7766 // if last effect in the chain, output samples to chain 7767 // output buffer, otherwise to chain input buffer 7768 if (idx_insert == size) { 7769 if (idx_insert != 0) { 7770 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7771 mEffects[idx_insert-1]->configure(); 7772 } 7773 effect->setOutBuffer(mOutBuffer); 7774 } else { 7775 effect->setOutBuffer(mInBuffer); 7776 } 7777 mEffects.insertAt(effect, idx_insert); 7778 7779 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7780 } 7781 effect->configure(); 7782 return NO_ERROR; 7783} 7784 7785// removeEffect_l() must be called with PlaybackThread::mLock held 7786size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7787{ 7788 Mutex::Autolock _l(mLock); 7789 size_t size = mEffects.size(); 7790 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7791 7792 for (size_t i = 0; i < size; i++) { 7793 if (effect == mEffects[i]) { 7794 // calling stop here will remove pre-processing effect from the audio HAL. 7795 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7796 // the middle of a read from audio HAL 7797 if (mEffects[i]->state() == EffectModule::ACTIVE || 7798 mEffects[i]->state() == EffectModule::STOPPING) { 7799 mEffects[i]->stop(); 7800 } 7801 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7802 delete[] effect->inBuffer(); 7803 } else { 7804 if (i == size - 1 && i != 0) { 7805 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7806 mEffects[i - 1]->configure(); 7807 } 7808 } 7809 mEffects.removeAt(i); 7810 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7811 break; 7812 } 7813 } 7814 7815 return mEffects.size(); 7816} 7817 7818// setDevice_l() must be called with PlaybackThread::mLock held 7819void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7820{ 7821 size_t size = mEffects.size(); 7822 for (size_t i = 0; i < size; i++) { 7823 mEffects[i]->setDevice(device); 7824 } 7825} 7826 7827// setMode_l() must be called with PlaybackThread::mLock held 7828void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7829{ 7830 size_t size = mEffects.size(); 7831 for (size_t i = 0; i < size; i++) { 7832 mEffects[i]->setMode(mode); 7833 } 7834} 7835 7836// setVolume_l() must be called with PlaybackThread::mLock held 7837bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7838{ 7839 uint32_t newLeft = *left; 7840 uint32_t newRight = *right; 7841 bool hasControl = false; 7842 int ctrlIdx = -1; 7843 size_t size = mEffects.size(); 7844 7845 // first update volume controller 7846 for (size_t i = size; i > 0; i--) { 7847 if (mEffects[i - 1]->isProcessEnabled() && 7848 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7849 ctrlIdx = i - 1; 7850 hasControl = true; 7851 break; 7852 } 7853 } 7854 7855 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7856 if (hasControl) { 7857 *left = mNewLeftVolume; 7858 *right = mNewRightVolume; 7859 } 7860 return hasControl; 7861 } 7862 7863 mVolumeCtrlIdx = ctrlIdx; 7864 mLeftVolume = newLeft; 7865 mRightVolume = newRight; 7866 7867 // second get volume update from volume controller 7868 if (ctrlIdx >= 0) { 7869 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7870 mNewLeftVolume = newLeft; 7871 mNewRightVolume = newRight; 7872 } 7873 // then indicate volume to all other effects in chain. 7874 // Pass altered volume to effects before volume controller 7875 // and requested volume to effects after controller 7876 uint32_t lVol = newLeft; 7877 uint32_t rVol = newRight; 7878 7879 for (size_t i = 0; i < size; i++) { 7880 if ((int)i == ctrlIdx) continue; 7881 // this also works for ctrlIdx == -1 when there is no volume controller 7882 if ((int)i > ctrlIdx) { 7883 lVol = *left; 7884 rVol = *right; 7885 } 7886 mEffects[i]->setVolume(&lVol, &rVol, false); 7887 } 7888 *left = newLeft; 7889 *right = newRight; 7890 7891 return hasControl; 7892} 7893 7894status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7895{ 7896 const size_t SIZE = 256; 7897 char buffer[SIZE]; 7898 String8 result; 7899 7900 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7901 result.append(buffer); 7902 7903 bool locked = tryLock(mLock); 7904 // failed to lock - AudioFlinger is probably deadlocked 7905 if (!locked) { 7906 result.append("\tCould not lock mutex:\n"); 7907 } 7908 7909 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7910 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7911 mEffects.size(), 7912 (uint32_t)mInBuffer, 7913 (uint32_t)mOutBuffer, 7914 mActiveTrackCnt); 7915 result.append(buffer); 7916 write(fd, result.string(), result.size()); 7917 7918 for (size_t i = 0; i < mEffects.size(); ++i) { 7919 sp<EffectModule> effect = mEffects[i]; 7920 if (effect != 0) { 7921 effect->dump(fd, args); 7922 } 7923 } 7924 7925 if (locked) { 7926 mLock.unlock(); 7927 } 7928 7929 return NO_ERROR; 7930} 7931 7932// must be called with ThreadBase::mLock held 7933void AudioFlinger::EffectChain::setEffectSuspended_l( 7934 const effect_uuid_t *type, bool suspend) 7935{ 7936 sp<SuspendedEffectDesc> desc; 7937 // use effect type UUID timelow as key as there is no real risk of identical 7938 // timeLow fields among effect type UUIDs. 7939 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7940 if (suspend) { 7941 if (index >= 0) { 7942 desc = mSuspendedEffects.valueAt(index); 7943 } else { 7944 desc = new SuspendedEffectDesc(); 7945 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7946 mSuspendedEffects.add(type->timeLow, desc); 7947 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7948 } 7949 if (desc->mRefCount++ == 0) { 7950 sp<EffectModule> effect = getEffectIfEnabled(type); 7951 if (effect != 0) { 7952 desc->mEffect = effect; 7953 effect->setSuspended(true); 7954 effect->setEnabled(false); 7955 } 7956 } 7957 } else { 7958 if (index < 0) { 7959 return; 7960 } 7961 desc = mSuspendedEffects.valueAt(index); 7962 if (desc->mRefCount <= 0) { 7963 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7964 desc->mRefCount = 1; 7965 } 7966 if (--desc->mRefCount == 0) { 7967 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7968 if (desc->mEffect != 0) { 7969 sp<EffectModule> effect = desc->mEffect.promote(); 7970 if (effect != 0) { 7971 effect->setSuspended(false); 7972 sp<EffectHandle> handle = effect->controlHandle(); 7973 if (handle != 0) { 7974 effect->setEnabled(handle->enabled()); 7975 } 7976 } 7977 desc->mEffect.clear(); 7978 } 7979 mSuspendedEffects.removeItemsAt(index); 7980 } 7981 } 7982} 7983 7984// must be called with ThreadBase::mLock held 7985void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7986{ 7987 sp<SuspendedEffectDesc> desc; 7988 7989 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7990 if (suspend) { 7991 if (index >= 0) { 7992 desc = mSuspendedEffects.valueAt(index); 7993 } else { 7994 desc = new SuspendedEffectDesc(); 7995 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7996 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7997 } 7998 if (desc->mRefCount++ == 0) { 7999 Vector< sp<EffectModule> > effects; 8000 getSuspendEligibleEffects(effects); 8001 for (size_t i = 0; i < effects.size(); i++) { 8002 setEffectSuspended_l(&effects[i]->desc().type, true); 8003 } 8004 } 8005 } else { 8006 if (index < 0) { 8007 return; 8008 } 8009 desc = mSuspendedEffects.valueAt(index); 8010 if (desc->mRefCount <= 0) { 8011 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8012 desc->mRefCount = 1; 8013 } 8014 if (--desc->mRefCount == 0) { 8015 Vector<const effect_uuid_t *> types; 8016 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8017 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8018 continue; 8019 } 8020 types.add(&mSuspendedEffects.valueAt(i)->mType); 8021 } 8022 for (size_t i = 0; i < types.size(); i++) { 8023 setEffectSuspended_l(types[i], false); 8024 } 8025 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8026 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8027 } 8028 } 8029} 8030 8031 8032// The volume effect is used for automated tests only 8033#ifndef OPENSL_ES_H_ 8034static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8035 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8036const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8037#endif //OPENSL_ES_H_ 8038 8039bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8040{ 8041 // auxiliary effects and visualizer are never suspended on output mix 8042 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8043 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8044 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8045 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8046 return false; 8047 } 8048 return true; 8049} 8050 8051void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8052{ 8053 effects.clear(); 8054 for (size_t i = 0; i < mEffects.size(); i++) { 8055 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8056 effects.add(mEffects[i]); 8057 } 8058 } 8059} 8060 8061sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8062 const effect_uuid_t *type) 8063{ 8064 sp<EffectModule> effect = getEffectFromType_l(type); 8065 return effect != 0 && effect->isEnabled() ? effect : 0; 8066} 8067 8068void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8069 bool enabled) 8070{ 8071 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8072 if (enabled) { 8073 if (index < 0) { 8074 // if the effect is not suspend check if all effects are suspended 8075 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8076 if (index < 0) { 8077 return; 8078 } 8079 if (!isEffectEligibleForSuspend(effect->desc())) { 8080 return; 8081 } 8082 setEffectSuspended_l(&effect->desc().type, enabled); 8083 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8084 if (index < 0) { 8085 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8086 return; 8087 } 8088 } 8089 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8090 effect->desc().type.timeLow); 8091 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8092 // if effect is requested to suspended but was not yet enabled, supend it now. 8093 if (desc->mEffect == 0) { 8094 desc->mEffect = effect; 8095 effect->setEnabled(false); 8096 effect->setSuspended(true); 8097 } 8098 } else { 8099 if (index < 0) { 8100 return; 8101 } 8102 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8103 effect->desc().type.timeLow); 8104 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8105 desc->mEffect.clear(); 8106 effect->setSuspended(false); 8107 } 8108} 8109 8110#undef LOG_TAG 8111#define LOG_TAG "AudioFlinger" 8112 8113// ---------------------------------------------------------------------------- 8114 8115status_t AudioFlinger::onTransact( 8116 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8117{ 8118 return BnAudioFlinger::onTransact(code, data, reply, flags); 8119} 8120 8121}; // namespace android 8122