AudioFlinger.cpp revision d3cee0b1f77baa4fb7a049eb757e9f5006890726
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51#include "ServiceUtilities.h" 52 53#include <media/EffectsFactoryApi.h> 54#include <audio_effects/effect_visualizer.h> 55#include <audio_effects/effect_ns.h> 56#include <audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <cpustats/ThreadCpuUsage.h> 61#include <powermanager/PowerManager.h> 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63 64#include <common_time/cc_helper.h> 65#include <common_time/local_clock.h> 66 67// ---------------------------------------------------------------------------- 68 69 70namespace android { 71 72static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74 75static const float MAX_GAIN = 4096.0f; 76static const uint32_t MAX_GAIN_INT = 0x1000; 77 78// retry counts for buffer fill timeout 79// 50 * ~20msecs = 1 second 80static const int8_t kMaxTrackRetries = 50; 81static const int8_t kMaxTrackStartupRetries = 50; 82// allow less retry attempts on direct output thread. 83// direct outputs can be a scarce resource in audio hardware and should 84// be released as quickly as possible. 85static const int8_t kMaxTrackRetriesDirect = 2; 86 87static const int kDumpLockRetries = 50; 88static const int kDumpLockSleepUs = 20000; 89 90// don't warn about blocked writes or record buffer overflows more often than this 91static const nsecs_t kWarningThrottleNs = seconds(5); 92 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_INIT; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_INIT; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid cnt\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 // FIXME dead, remove from IAudioFlinger 436 uint32_t flags, 437 const sp<IMemory>& sharedBuffer, 438 audio_io_handle_t output, 439 bool isTimed, 440 int *sessionId, 441 status_t *status) 442{ 443 sp<PlaybackThread::Track> track; 444 sp<TrackHandle> trackHandle; 445 sp<Client> client; 446 status_t lStatus; 447 int lSessionId; 448 449 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 450 // but if someone uses binder directly they could bypass that and cause us to crash 451 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 452 ALOGE("createTrack() invalid stream type %d", streamType); 453 lStatus = BAD_VALUE; 454 goto Exit; 455 } 456 457 { 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 PlaybackThread *effectThread = NULL; 461 if (thread == NULL) { 462 ALOGE("unknown output thread"); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 client = registerPid_l(pid); 468 469 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 470 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 471 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 472 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 473 if (mPlaybackThreads.keyAt(i) != output) { 474 // prevent same audio session on different output threads 475 uint32_t sessions = t->hasAudioSession(*sessionId); 476 if (sessions & PlaybackThread::TRACK_SESSION) { 477 ALOGE("createTrack() session ID %d already in use", *sessionId); 478 lStatus = BAD_VALUE; 479 goto Exit; 480 } 481 // check if an effect with same session ID is waiting for a track to be created 482 if (sessions & PlaybackThread::EFFECT_SESSION) { 483 effectThread = t.get(); 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 } 508 if (lStatus == NO_ERROR) { 509 trackHandle = new TrackHandle(track); 510 } else { 511 // remove local strong reference to Client before deleting the Track so that the Client 512 // destructor is called by the TrackBase destructor with mLock held 513 client.clear(); 514 track.clear(); 515 } 516 517Exit: 518 if(status) { 519 *status = lStatus; 520 } 521 return trackHandle; 522} 523 524uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 525{ 526 Mutex::Autolock _l(mLock); 527 PlaybackThread *thread = checkPlaybackThread_l(output); 528 if (thread == NULL) { 529 ALOGW("sampleRate() unknown thread %d", output); 530 return 0; 531 } 532 return thread->sampleRate(); 533} 534 535int AudioFlinger::channelCount(audio_io_handle_t output) const 536{ 537 Mutex::Autolock _l(mLock); 538 PlaybackThread *thread = checkPlaybackThread_l(output); 539 if (thread == NULL) { 540 ALOGW("channelCount() unknown thread %d", output); 541 return 0; 542 } 543 return thread->channelCount(); 544} 545 546audio_format_t AudioFlinger::format(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("format() unknown thread %d", output); 552 return AUDIO_FORMAT_INVALID; 553 } 554 return thread->format(); 555} 556 557size_t AudioFlinger::frameCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("frameCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->frameCount(); 566} 567 568uint32_t AudioFlinger::latency(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("latency() unknown thread %d", output); 574 return 0; 575 } 576 return thread->latency(); 577} 578 579status_t AudioFlinger::setMasterVolume(float value) 580{ 581 status_t ret = initCheck(); 582 if (ret != NO_ERROR) { 583 return ret; 584 } 585 586 // check calling permissions 587 if (!settingsAllowed()) { 588 return PERMISSION_DENIED; 589 } 590 591 float swmv = value; 592 593 // when hw supports master volume, don't scale in sw mixer 594 if (MVS_NONE != mMasterVolumeSupportLvl) { 595 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 596 AutoMutex lock(mHardwareLock); 597 audio_hw_device_t *dev = mAudioHwDevs[i]; 598 599 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 600 if (NULL != dev->set_master_volume) { 601 dev->set_master_volume(dev, value); 602 } 603 mHardwareStatus = AUDIO_HW_IDLE; 604 } 605 606 swmv = 1.0; 607 } 608 609 Mutex::Autolock _l(mLock); 610 mMasterVolume = value; 611 mMasterVolumeSW = swmv; 612 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 613 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 614 615 return NO_ERROR; 616} 617 618status_t AudioFlinger::setMode(audio_mode_t mode) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 630 ALOGW("Illegal value: setMode(%d)", mode); 631 return BAD_VALUE; 632 } 633 634 { // scope for the lock 635 AutoMutex lock(mHardwareLock); 636 mHardwareStatus = AUDIO_HW_SET_MODE; 637 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 if (NO_ERROR == ret) { 642 Mutex::Autolock _l(mLock); 643 mMode = mode; 644 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 645 mPlaybackThreads.valueAt(i)->setMode(mode); 646 } 647 648 return ret; 649} 650 651status_t AudioFlinger::setMicMute(bool state) 652{ 653 status_t ret = initCheck(); 654 if (ret != NO_ERROR) { 655 return ret; 656 } 657 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 AutoMutex lock(mHardwareLock); 664 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 665 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 666 mHardwareStatus = AUDIO_HW_IDLE; 667 return ret; 668} 669 670bool AudioFlinger::getMicMute() const 671{ 672 status_t ret = initCheck(); 673 if (ret != NO_ERROR) { 674 return false; 675 } 676 677 bool state = AUDIO_MODE_INVALID; 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 680 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return state; 683} 684 685status_t AudioFlinger::setMasterMute(bool muted) 686{ 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 Mutex::Autolock _l(mLock); 693 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 694 mMasterMute = muted; 695 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 696 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 697 698 return NO_ERROR; 699} 700 701float AudioFlinger::masterVolume() const 702{ 703 Mutex::Autolock _l(mLock); 704 return masterVolume_l(); 705} 706 707float AudioFlinger::masterVolumeSW() const 708{ 709 Mutex::Autolock _l(mLock); 710 return masterVolumeSW_l(); 711} 712 713bool AudioFlinger::masterMute() const 714{ 715 Mutex::Autolock _l(mLock); 716 return masterMute_l(); 717} 718 719float AudioFlinger::masterVolume_l() const 720{ 721 if (MVS_FULL == mMasterVolumeSupportLvl) { 722 float ret_val; 723 AutoMutex lock(mHardwareLock); 724 725 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 726 assert(NULL != mPrimaryHardwareDev); 727 assert(NULL != mPrimaryHardwareDev->get_master_volume); 728 729 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 730 mHardwareStatus = AUDIO_HW_IDLE; 731 return ret_val; 732 } 733 734 return mMasterVolume; 735} 736 737status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 738 audio_io_handle_t output) 739{ 740 // check calling permissions 741 if (!settingsAllowed()) { 742 return PERMISSION_DENIED; 743 } 744 745 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 746 ALOGE("setStreamVolume() invalid stream %d", stream); 747 return BAD_VALUE; 748 } 749 750 AutoMutex lock(mLock); 751 PlaybackThread *thread = NULL; 752 if (output) { 753 thread = checkPlaybackThread_l(output); 754 if (thread == NULL) { 755 return BAD_VALUE; 756 } 757 } 758 759 mStreamTypes[stream].volume = value; 760 761 if (thread == NULL) { 762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 763 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 764 } 765 } else { 766 thread->setStreamVolume(stream, value); 767 } 768 769 return NO_ERROR; 770} 771 772status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 773{ 774 // check calling permissions 775 if (!settingsAllowed()) { 776 return PERMISSION_DENIED; 777 } 778 779 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 780 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 781 ALOGE("setStreamMute() invalid stream %d", stream); 782 return BAD_VALUE; 783 } 784 785 AutoMutex lock(mLock); 786 mStreamTypes[stream].mute = muted; 787 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 788 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 789 790 return NO_ERROR; 791} 792 793float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 794{ 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 796 return 0.0f; 797 } 798 799 AutoMutex lock(mLock); 800 float volume; 801 if (output) { 802 PlaybackThread *thread = checkPlaybackThread_l(output); 803 if (thread == NULL) { 804 return 0.0f; 805 } 806 volume = thread->streamVolume(stream); 807 } else { 808 volume = streamVolume_l(stream); 809 } 810 811 return volume; 812} 813 814bool AudioFlinger::streamMute(audio_stream_type_t stream) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return true; 818 } 819 820 AutoMutex lock(mLock); 821 return streamMute_l(stream); 822} 823 824status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 825{ 826 status_t result; 827 828 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 829 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 830 // check calling permissions 831 if (!settingsAllowed()) { 832 return PERMISSION_DENIED; 833 } 834 835 // ioHandle == 0 means the parameters are global to the audio hardware interface 836 if (ioHandle == 0) { 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_SET_PARAMETER; 839 status_t final_result = NO_ERROR; 840 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 841 audio_hw_device_t *dev = mAudioHwDevs[i]; 842 result = dev->set_parameters(dev, keyValuePairs.string()); 843 final_result = result ?: final_result; 844 } 845 mHardwareStatus = AUDIO_HW_IDLE; 846 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 847 AudioParameter param = AudioParameter(keyValuePairs); 848 String8 value; 849 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 850 Mutex::Autolock _l(mLock); 851 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 852 if (mBtNrecIsOff != btNrecIsOff) { 853 for (size_t i = 0; i < mRecordThreads.size(); i++) { 854 sp<RecordThread> thread = mRecordThreads.valueAt(i); 855 RecordThread::RecordTrack *track = thread->track(); 856 if (track != NULL) { 857 audio_devices_t device = (audio_devices_t)( 858 thread->device() & AUDIO_DEVICE_IN_ALL); 859 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 860 thread->setEffectSuspended(FX_IID_AEC, 861 suspend, 862 track->sessionId()); 863 thread->setEffectSuspended(FX_IID_NS, 864 suspend, 865 track->sessionId()); 866 } 867 } 868 mBtNrecIsOff = btNrecIsOff; 869 } 870 } 871 return final_result; 872 } 873 874 // hold a strong ref on thread in case closeOutput() or closeInput() is called 875 // and the thread is exited once the lock is released 876 sp<ThreadBase> thread; 877 { 878 Mutex::Autolock _l(mLock); 879 thread = checkPlaybackThread_l(ioHandle); 880 if (thread == NULL) { 881 thread = checkRecordThread_l(ioHandle); 882 } else if (thread == primaryPlaybackThread_l()) { 883 // indicate output device change to all input threads for pre processing 884 AudioParameter param = AudioParameter(keyValuePairs); 885 int value; 886 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 887 for (size_t i = 0; i < mRecordThreads.size(); i++) { 888 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 889 } 890 } 891 } 892 } 893 if (thread != 0) { 894 return thread->setParameters(keyValuePairs); 895 } 896 return BAD_VALUE; 897} 898 899String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 900{ 901// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 902// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 903 904 if (ioHandle == 0) { 905 String8 out_s8; 906 907 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 908 audio_hw_device_t *dev = mAudioHwDevs[i]; 909 char *s = dev->get_parameters(dev, keys.string()); 910 out_s8 += String8(s ? s : ""); 911 free(s); 912 } 913 return out_s8; 914 } 915 916 Mutex::Autolock _l(mLock); 917 918 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 919 if (playbackThread != NULL) { 920 return playbackThread->getParameters(keys); 921 } 922 RecordThread *recordThread = checkRecordThread_l(ioHandle); 923 if (recordThread != NULL) { 924 return recordThread->getParameters(keys); 925 } 926 return String8(""); 927} 928 929size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 930{ 931 status_t ret = initCheck(); 932 if (ret != NO_ERROR) { 933 return 0; 934 } 935 936 AutoMutex lock(mHardwareLock); 937 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 938 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 939 mHardwareStatus = AUDIO_HW_IDLE; 940 return size; 941} 942 943unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 944{ 945 if (ioHandle == 0) { 946 return 0; 947 } 948 949 Mutex::Autolock _l(mLock); 950 951 RecordThread *recordThread = checkRecordThread_l(ioHandle); 952 if (recordThread != NULL) { 953 return recordThread->getInputFramesLost(); 954 } 955 return 0; 956} 957 958status_t AudioFlinger::setVoiceVolume(float value) 959{ 960 status_t ret = initCheck(); 961 if (ret != NO_ERROR) { 962 return ret; 963 } 964 965 // check calling permissions 966 if (!settingsAllowed()) { 967 return PERMISSION_DENIED; 968 } 969 970 AutoMutex lock(mHardwareLock); 971 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 972 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 973 mHardwareStatus = AUDIO_HW_IDLE; 974 975 return ret; 976} 977 978status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 979 audio_io_handle_t output) const 980{ 981 status_t status; 982 983 Mutex::Autolock _l(mLock); 984 985 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 986 if (playbackThread != NULL) { 987 return playbackThread->getRenderPosition(halFrames, dspFrames); 988 } 989 990 return BAD_VALUE; 991} 992 993void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 994{ 995 996 Mutex::Autolock _l(mLock); 997 998 pid_t pid = IPCThreadState::self()->getCallingPid(); 999 if (mNotificationClients.indexOfKey(pid) < 0) { 1000 sp<NotificationClient> notificationClient = new NotificationClient(this, 1001 client, 1002 pid); 1003 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1004 1005 mNotificationClients.add(pid, notificationClient); 1006 1007 sp<IBinder> binder = client->asBinder(); 1008 binder->linkToDeath(notificationClient); 1009 1010 // the config change is always sent from playback or record threads to avoid deadlock 1011 // with AudioSystem::gLock 1012 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1013 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1014 } 1015 1016 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1017 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1018 } 1019 } 1020} 1021 1022void AudioFlinger::removeNotificationClient(pid_t pid) 1023{ 1024 Mutex::Autolock _l(mLock); 1025 1026 ssize_t index = mNotificationClients.indexOfKey(pid); 1027 if (index >= 0) { 1028 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 1029 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 1030 mNotificationClients.removeItem(pid); 1031 } 1032 1033 ALOGV("%d died, releasing its sessions", pid); 1034 size_t num = mAudioSessionRefs.size(); 1035 bool removed = false; 1036 for (size_t i = 0; i< num; ) { 1037 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1038 ALOGV(" pid %d @ %d", ref->pid, i); 1039 if (ref->pid == pid) { 1040 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1041 mAudioSessionRefs.removeAt(i); 1042 delete ref; 1043 removed = true; 1044 num--; 1045 } else { 1046 i++; 1047 } 1048 } 1049 if (removed) { 1050 purgeStaleEffects_l(); 1051 } 1052} 1053 1054// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1055void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1056{ 1057 size_t size = mNotificationClients.size(); 1058 for (size_t i = 0; i < size; i++) { 1059 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1060 param2); 1061 } 1062} 1063 1064// removeClient_l() must be called with AudioFlinger::mLock held 1065void AudioFlinger::removeClient_l(pid_t pid) 1066{ 1067 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1068 mClients.removeItem(pid); 1069} 1070 1071 1072// ---------------------------------------------------------------------------- 1073 1074AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1075 uint32_t device, type_t type) 1076 : Thread(false), 1077 mType(type), 1078 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1079 // mChannelMask 1080 mChannelCount(0), 1081 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1082 mParamStatus(NO_ERROR), 1083 mStandby(false), mId(id), 1084 mDevice(device), 1085 mDeathRecipient(new PMDeathRecipient(this)) 1086{ 1087} 1088 1089AudioFlinger::ThreadBase::~ThreadBase() 1090{ 1091 mParamCond.broadcast(); 1092 // do not lock the mutex in destructor 1093 releaseWakeLock_l(); 1094 if (mPowerManager != 0) { 1095 sp<IBinder> binder = mPowerManager->asBinder(); 1096 binder->unlinkToDeath(mDeathRecipient); 1097 } 1098} 1099 1100void AudioFlinger::ThreadBase::exit() 1101{ 1102 ALOGV("ThreadBase::exit"); 1103 { 1104 // This lock prevents the following race in thread (uniprocessor for illustration): 1105 // if (!exitPending()) { 1106 // // context switch from here to exit() 1107 // // exit() calls requestExit(), what exitPending() observes 1108 // // exit() calls signal(), which is dropped since no waiters 1109 // // context switch back from exit() to here 1110 // mWaitWorkCV.wait(...); 1111 // // now thread is hung 1112 // } 1113 AutoMutex lock(mLock); 1114 requestExit(); 1115 mWaitWorkCV.signal(); 1116 } 1117 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1118 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1119 requestExitAndWait(); 1120} 1121 1122status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1123{ 1124 status_t status; 1125 1126 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1127 Mutex::Autolock _l(mLock); 1128 1129 mNewParameters.add(keyValuePairs); 1130 mWaitWorkCV.signal(); 1131 // wait condition with timeout in case the thread loop has exited 1132 // before the request could be processed 1133 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1134 status = mParamStatus; 1135 mWaitWorkCV.signal(); 1136 } else { 1137 status = TIMED_OUT; 1138 } 1139 return status; 1140} 1141 1142void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1143{ 1144 Mutex::Autolock _l(mLock); 1145 sendConfigEvent_l(event, param); 1146} 1147 1148// sendConfigEvent_l() must be called with ThreadBase::mLock held 1149void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1150{ 1151 ConfigEvent configEvent; 1152 configEvent.mEvent = event; 1153 configEvent.mParam = param; 1154 mConfigEvents.add(configEvent); 1155 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1156 mWaitWorkCV.signal(); 1157} 1158 1159void AudioFlinger::ThreadBase::processConfigEvents() 1160{ 1161 mLock.lock(); 1162 while(!mConfigEvents.isEmpty()) { 1163 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1164 ConfigEvent configEvent = mConfigEvents[0]; 1165 mConfigEvents.removeAt(0); 1166 // release mLock before locking AudioFlinger mLock: lock order is always 1167 // AudioFlinger then ThreadBase to avoid cross deadlock 1168 mLock.unlock(); 1169 mAudioFlinger->mLock.lock(); 1170 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1171 mAudioFlinger->mLock.unlock(); 1172 mLock.lock(); 1173 } 1174 mLock.unlock(); 1175} 1176 1177status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1178{ 1179 const size_t SIZE = 256; 1180 char buffer[SIZE]; 1181 String8 result; 1182 1183 bool locked = tryLock(mLock); 1184 if (!locked) { 1185 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1186 write(fd, buffer, strlen(buffer)); 1187 } 1188 1189 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1190 result.append(buffer); 1191 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1192 result.append(buffer); 1193 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1194 result.append(buffer); 1195 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1196 result.append(buffer); 1197 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1198 result.append(buffer); 1199 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1200 result.append(buffer); 1201 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1202 result.append(buffer); 1203 1204 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1205 result.append(buffer); 1206 result.append(" Index Command"); 1207 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1208 snprintf(buffer, SIZE, "\n %02d ", i); 1209 result.append(buffer); 1210 result.append(mNewParameters[i]); 1211 } 1212 1213 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1214 result.append(buffer); 1215 snprintf(buffer, SIZE, " Index event param\n"); 1216 result.append(buffer); 1217 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1218 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1219 result.append(buffer); 1220 } 1221 result.append("\n"); 1222 1223 write(fd, result.string(), result.size()); 1224 1225 if (locked) { 1226 mLock.unlock(); 1227 } 1228 return NO_ERROR; 1229} 1230 1231status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1232{ 1233 const size_t SIZE = 256; 1234 char buffer[SIZE]; 1235 String8 result; 1236 1237 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1238 write(fd, buffer, strlen(buffer)); 1239 1240 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1241 sp<EffectChain> chain = mEffectChains[i]; 1242 if (chain != 0) { 1243 chain->dump(fd, args); 1244 } 1245 } 1246 return NO_ERROR; 1247} 1248 1249void AudioFlinger::ThreadBase::acquireWakeLock() 1250{ 1251 Mutex::Autolock _l(mLock); 1252 acquireWakeLock_l(); 1253} 1254 1255void AudioFlinger::ThreadBase::acquireWakeLock_l() 1256{ 1257 if (mPowerManager == 0) { 1258 // use checkService() to avoid blocking if power service is not up yet 1259 sp<IBinder> binder = 1260 defaultServiceManager()->checkService(String16("power")); 1261 if (binder == 0) { 1262 ALOGW("Thread %s cannot connect to the power manager service", mName); 1263 } else { 1264 mPowerManager = interface_cast<IPowerManager>(binder); 1265 binder->linkToDeath(mDeathRecipient); 1266 } 1267 } 1268 if (mPowerManager != 0) { 1269 sp<IBinder> binder = new BBinder(); 1270 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1271 binder, 1272 String16(mName)); 1273 if (status == NO_ERROR) { 1274 mWakeLockToken = binder; 1275 } 1276 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1277 } 1278} 1279 1280void AudioFlinger::ThreadBase::releaseWakeLock() 1281{ 1282 Mutex::Autolock _l(mLock); 1283 releaseWakeLock_l(); 1284} 1285 1286void AudioFlinger::ThreadBase::releaseWakeLock_l() 1287{ 1288 if (mWakeLockToken != 0) { 1289 ALOGV("releaseWakeLock_l() %s", mName); 1290 if (mPowerManager != 0) { 1291 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1292 } 1293 mWakeLockToken.clear(); 1294 } 1295} 1296 1297void AudioFlinger::ThreadBase::clearPowerManager() 1298{ 1299 Mutex::Autolock _l(mLock); 1300 releaseWakeLock_l(); 1301 mPowerManager.clear(); 1302} 1303 1304void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1305{ 1306 sp<ThreadBase> thread = mThread.promote(); 1307 if (thread != 0) { 1308 thread->clearPowerManager(); 1309 } 1310 ALOGW("power manager service died !!!"); 1311} 1312 1313void AudioFlinger::ThreadBase::setEffectSuspended( 1314 const effect_uuid_t *type, bool suspend, int sessionId) 1315{ 1316 Mutex::Autolock _l(mLock); 1317 setEffectSuspended_l(type, suspend, sessionId); 1318} 1319 1320void AudioFlinger::ThreadBase::setEffectSuspended_l( 1321 const effect_uuid_t *type, bool suspend, int sessionId) 1322{ 1323 sp<EffectChain> chain = getEffectChain_l(sessionId); 1324 if (chain != 0) { 1325 if (type != NULL) { 1326 chain->setEffectSuspended_l(type, suspend); 1327 } else { 1328 chain->setEffectSuspendedAll_l(suspend); 1329 } 1330 } 1331 1332 updateSuspendedSessions_l(type, suspend, sessionId); 1333} 1334 1335void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1336{ 1337 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1338 if (index < 0) { 1339 return; 1340 } 1341 1342 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1343 mSuspendedSessions.editValueAt(index); 1344 1345 for (size_t i = 0; i < sessionEffects.size(); i++) { 1346 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1347 for (int j = 0; j < desc->mRefCount; j++) { 1348 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1349 chain->setEffectSuspendedAll_l(true); 1350 } else { 1351 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1352 desc->mType.timeLow); 1353 chain->setEffectSuspended_l(&desc->mType, true); 1354 } 1355 } 1356 } 1357} 1358 1359void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1360 bool suspend, 1361 int sessionId) 1362{ 1363 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1364 1365 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1366 1367 if (suspend) { 1368 if (index >= 0) { 1369 sessionEffects = mSuspendedSessions.editValueAt(index); 1370 } else { 1371 mSuspendedSessions.add(sessionId, sessionEffects); 1372 } 1373 } else { 1374 if (index < 0) { 1375 return; 1376 } 1377 sessionEffects = mSuspendedSessions.editValueAt(index); 1378 } 1379 1380 1381 int key = EffectChain::kKeyForSuspendAll; 1382 if (type != NULL) { 1383 key = type->timeLow; 1384 } 1385 index = sessionEffects.indexOfKey(key); 1386 1387 sp <SuspendedSessionDesc> desc; 1388 if (suspend) { 1389 if (index >= 0) { 1390 desc = sessionEffects.valueAt(index); 1391 } else { 1392 desc = new SuspendedSessionDesc(); 1393 if (type != NULL) { 1394 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1395 } 1396 sessionEffects.add(key, desc); 1397 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1398 } 1399 desc->mRefCount++; 1400 } else { 1401 if (index < 0) { 1402 return; 1403 } 1404 desc = sessionEffects.valueAt(index); 1405 if (--desc->mRefCount == 0) { 1406 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1407 sessionEffects.removeItemsAt(index); 1408 if (sessionEffects.isEmpty()) { 1409 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1410 sessionId); 1411 mSuspendedSessions.removeItem(sessionId); 1412 } 1413 } 1414 } 1415 if (!sessionEffects.isEmpty()) { 1416 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1417 } 1418} 1419 1420void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1421 bool enabled, 1422 int sessionId) 1423{ 1424 Mutex::Autolock _l(mLock); 1425 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1426} 1427 1428void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1429 bool enabled, 1430 int sessionId) 1431{ 1432 if (mType != RECORD) { 1433 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1434 // another session. This gives the priority to well behaved effect control panels 1435 // and applications not using global effects. 1436 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1437 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1438 } 1439 } 1440 1441 sp<EffectChain> chain = getEffectChain_l(sessionId); 1442 if (chain != 0) { 1443 chain->checkSuspendOnEffectEnabled(effect, enabled); 1444 } 1445} 1446 1447// ---------------------------------------------------------------------------- 1448 1449AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1450 AudioStreamOut* output, 1451 audio_io_handle_t id, 1452 uint32_t device, 1453 type_t type) 1454 : ThreadBase(audioFlinger, id, device, type), 1455 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1456 // Assumes constructor is called by AudioFlinger with it's mLock held, 1457 // but it would be safer to explicitly pass initial masterMute as parameter 1458 mMasterMute(audioFlinger->masterMute_l()), 1459 // mStreamTypes[] initialized in constructor body 1460 mOutput(output), 1461 // Assumes constructor is called by AudioFlinger with it's mLock held, 1462 // but it would be safer to explicitly pass initial masterVolume as parameter 1463 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1464 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1465{ 1466 snprintf(mName, kNameLength, "AudioOut_%d", id); 1467 1468 readOutputParameters(); 1469 1470 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1471 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1472 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1473 stream = (audio_stream_type_t) (stream + 1)) { 1474 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1475 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1476 // initialized by stream_type_t default constructor 1477 // mStreamTypes[stream].valid = true; 1478 } 1479 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1480 // because mAudioFlinger doesn't have one to copy from 1481} 1482 1483AudioFlinger::PlaybackThread::~PlaybackThread() 1484{ 1485 delete [] mMixBuffer; 1486} 1487 1488status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1489{ 1490 dumpInternals(fd, args); 1491 dumpTracks(fd, args); 1492 dumpEffectChains(fd, args); 1493 return NO_ERROR; 1494} 1495 1496status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1497{ 1498 const size_t SIZE = 256; 1499 char buffer[SIZE]; 1500 String8 result; 1501 1502 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1503 result.append(buffer); 1504 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1505 for (size_t i = 0; i < mTracks.size(); ++i) { 1506 sp<Track> track = mTracks[i]; 1507 if (track != 0) { 1508 track->dump(buffer, SIZE); 1509 result.append(buffer); 1510 } 1511 } 1512 1513 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1514 result.append(buffer); 1515 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1516 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1517 sp<Track> track = mActiveTracks[i].promote(); 1518 if (track != 0) { 1519 track->dump(buffer, SIZE); 1520 result.append(buffer); 1521 } 1522 } 1523 write(fd, result.string(), result.size()); 1524 return NO_ERROR; 1525} 1526 1527status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1528{ 1529 const size_t SIZE = 256; 1530 char buffer[SIZE]; 1531 String8 result; 1532 1533 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1534 result.append(buffer); 1535 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1536 result.append(buffer); 1537 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1538 result.append(buffer); 1539 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1540 result.append(buffer); 1541 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1542 result.append(buffer); 1543 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1544 result.append(buffer); 1545 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1546 result.append(buffer); 1547 write(fd, result.string(), result.size()); 1548 1549 dumpBase(fd, args); 1550 1551 return NO_ERROR; 1552} 1553 1554// Thread virtuals 1555status_t AudioFlinger::PlaybackThread::readyToRun() 1556{ 1557 status_t status = initCheck(); 1558 if (status == NO_ERROR) { 1559 ALOGI("AudioFlinger's thread %p ready to run", this); 1560 } else { 1561 ALOGE("No working audio driver found."); 1562 } 1563 return status; 1564} 1565 1566void AudioFlinger::PlaybackThread::onFirstRef() 1567{ 1568 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1569} 1570 1571// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1572sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1573 const sp<AudioFlinger::Client>& client, 1574 audio_stream_type_t streamType, 1575 uint32_t sampleRate, 1576 audio_format_t format, 1577 uint32_t channelMask, 1578 int frameCount, 1579 const sp<IMemory>& sharedBuffer, 1580 int sessionId, 1581 bool isTimed, 1582 status_t *status) 1583{ 1584 sp<Track> track; 1585 status_t lStatus; 1586 1587 if (mType == DIRECT) { 1588 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1589 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1590 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1591 "for output %p with format %d", 1592 sampleRate, format, channelMask, mOutput, mFormat); 1593 lStatus = BAD_VALUE; 1594 goto Exit; 1595 } 1596 } 1597 } else { 1598 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1599 if (sampleRate > mSampleRate*2) { 1600 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1601 lStatus = BAD_VALUE; 1602 goto Exit; 1603 } 1604 } 1605 1606 lStatus = initCheck(); 1607 if (lStatus != NO_ERROR) { 1608 ALOGE("Audio driver not initialized."); 1609 goto Exit; 1610 } 1611 1612 { // scope for mLock 1613 Mutex::Autolock _l(mLock); 1614 1615 // all tracks in same audio session must share the same routing strategy otherwise 1616 // conflicts will happen when tracks are moved from one output to another by audio policy 1617 // manager 1618 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1619 for (size_t i = 0; i < mTracks.size(); ++i) { 1620 sp<Track> t = mTracks[i]; 1621 if (t != 0) { 1622 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1623 if (sessionId == t->sessionId() && strategy != actual) { 1624 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1625 strategy, actual); 1626 lStatus = BAD_VALUE; 1627 goto Exit; 1628 } 1629 } 1630 } 1631 1632 if (!isTimed) { 1633 track = new Track(this, client, streamType, sampleRate, format, 1634 channelMask, frameCount, sharedBuffer, sessionId); 1635 } else { 1636 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1637 channelMask, frameCount, sharedBuffer, sessionId); 1638 } 1639 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1640 lStatus = NO_MEMORY; 1641 goto Exit; 1642 } 1643 mTracks.add(track); 1644 1645 sp<EffectChain> chain = getEffectChain_l(sessionId); 1646 if (chain != 0) { 1647 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1648 track->setMainBuffer(chain->inBuffer()); 1649 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1650 chain->incTrackCnt(); 1651 } 1652 1653 // invalidate track immediately if the stream type was moved to another thread since 1654 // createTrack() was called by the client process. 1655 if (!mStreamTypes[streamType].valid) { 1656 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1657 this, streamType); 1658 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1659 } 1660 } 1661 lStatus = NO_ERROR; 1662 1663Exit: 1664 if(status) { 1665 *status = lStatus; 1666 } 1667 return track; 1668} 1669 1670uint32_t AudioFlinger::PlaybackThread::latency() const 1671{ 1672 Mutex::Autolock _l(mLock); 1673 if (initCheck() == NO_ERROR) { 1674 return mOutput->stream->get_latency(mOutput->stream); 1675 } else { 1676 return 0; 1677 } 1678} 1679 1680void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1681{ 1682 Mutex::Autolock _l(mLock); 1683 mMasterVolume = value; 1684} 1685 1686void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1687{ 1688 Mutex::Autolock _l(mLock); 1689 setMasterMute_l(muted); 1690} 1691 1692void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1693{ 1694 Mutex::Autolock _l(mLock); 1695 mStreamTypes[stream].volume = value; 1696} 1697 1698void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1699{ 1700 Mutex::Autolock _l(mLock); 1701 mStreamTypes[stream].mute = muted; 1702} 1703 1704float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1705{ 1706 Mutex::Autolock _l(mLock); 1707 return mStreamTypes[stream].volume; 1708} 1709 1710// addTrack_l() must be called with ThreadBase::mLock held 1711status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1712{ 1713 status_t status = ALREADY_EXISTS; 1714 1715 // set retry count for buffer fill 1716 track->mRetryCount = kMaxTrackStartupRetries; 1717 if (mActiveTracks.indexOf(track) < 0) { 1718 // the track is newly added, make sure it fills up all its 1719 // buffers before playing. This is to ensure the client will 1720 // effectively get the latency it requested. 1721 track->mFillingUpStatus = Track::FS_FILLING; 1722 track->mResetDone = false; 1723 mActiveTracks.add(track); 1724 if (track->mainBuffer() != mMixBuffer) { 1725 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1726 if (chain != 0) { 1727 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1728 chain->incActiveTrackCnt(); 1729 } 1730 } 1731 1732 status = NO_ERROR; 1733 } 1734 1735 ALOGV("mWaitWorkCV.broadcast"); 1736 mWaitWorkCV.broadcast(); 1737 1738 return status; 1739} 1740 1741// destroyTrack_l() must be called with ThreadBase::mLock held 1742void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1743{ 1744 track->mState = TrackBase::TERMINATED; 1745 if (mActiveTracks.indexOf(track) < 0) { 1746 removeTrack_l(track); 1747 } 1748} 1749 1750void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1751{ 1752 mTracks.remove(track); 1753 deleteTrackName_l(track->name()); 1754 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1755 if (chain != 0) { 1756 chain->decTrackCnt(); 1757 } 1758} 1759 1760String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1761{ 1762 String8 out_s8 = String8(""); 1763 char *s; 1764 1765 Mutex::Autolock _l(mLock); 1766 if (initCheck() != NO_ERROR) { 1767 return out_s8; 1768 } 1769 1770 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1771 out_s8 = String8(s); 1772 free(s); 1773 return out_s8; 1774} 1775 1776// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1777void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1778 AudioSystem::OutputDescriptor desc; 1779 void *param2 = NULL; 1780 1781 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1782 1783 switch (event) { 1784 case AudioSystem::OUTPUT_OPENED: 1785 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1786 desc.channels = mChannelMask; 1787 desc.samplingRate = mSampleRate; 1788 desc.format = mFormat; 1789 desc.frameCount = mFrameCount; 1790 desc.latency = latency(); 1791 param2 = &desc; 1792 break; 1793 1794 case AudioSystem::STREAM_CONFIG_CHANGED: 1795 param2 = ¶m; 1796 case AudioSystem::OUTPUT_CLOSED: 1797 default: 1798 break; 1799 } 1800 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1801} 1802 1803void AudioFlinger::PlaybackThread::readOutputParameters() 1804{ 1805 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1806 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1807 mChannelCount = (uint16_t)popcount(mChannelMask); 1808 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1809 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1810 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1811 1812 // FIXME - Current mixer implementation only supports stereo output: Always 1813 // Allocate a stereo buffer even if HW output is mono. 1814 delete[] mMixBuffer; 1815 mMixBuffer = new int16_t[mFrameCount * 2]; 1816 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1817 1818 // force reconfiguration of effect chains and engines to take new buffer size and audio 1819 // parameters into account 1820 // Note that mLock is not held when readOutputParameters() is called from the constructor 1821 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1822 // matter. 1823 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1824 Vector< sp<EffectChain> > effectChains = mEffectChains; 1825 for (size_t i = 0; i < effectChains.size(); i ++) { 1826 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1827 } 1828} 1829 1830status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1831{ 1832 if (halFrames == NULL || dspFrames == NULL) { 1833 return BAD_VALUE; 1834 } 1835 Mutex::Autolock _l(mLock); 1836 if (initCheck() != NO_ERROR) { 1837 return INVALID_OPERATION; 1838 } 1839 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1840 1841 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1842} 1843 1844uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1845{ 1846 Mutex::Autolock _l(mLock); 1847 uint32_t result = 0; 1848 if (getEffectChain_l(sessionId) != 0) { 1849 result = EFFECT_SESSION; 1850 } 1851 1852 for (size_t i = 0; i < mTracks.size(); ++i) { 1853 sp<Track> track = mTracks[i]; 1854 if (sessionId == track->sessionId() && 1855 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1856 result |= TRACK_SESSION; 1857 break; 1858 } 1859 } 1860 1861 return result; 1862} 1863 1864uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1865{ 1866 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1867 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1868 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1869 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1870 } 1871 for (size_t i = 0; i < mTracks.size(); i++) { 1872 sp<Track> track = mTracks[i]; 1873 if (sessionId == track->sessionId() && 1874 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1875 return AudioSystem::getStrategyForStream(track->streamType()); 1876 } 1877 } 1878 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1879} 1880 1881 1882AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1883{ 1884 Mutex::Autolock _l(mLock); 1885 return mOutput; 1886} 1887 1888AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1889{ 1890 Mutex::Autolock _l(mLock); 1891 AudioStreamOut *output = mOutput; 1892 mOutput = NULL; 1893 return output; 1894} 1895 1896// this method must always be called either with ThreadBase mLock held or inside the thread loop 1897audio_stream_t* AudioFlinger::PlaybackThread::stream() 1898{ 1899 if (mOutput == NULL) { 1900 return NULL; 1901 } 1902 return &mOutput->stream->common; 1903} 1904 1905uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1906{ 1907 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1908 // decoding and transfer time. So sleeping for half of the latency would likely cause 1909 // underruns 1910 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1911 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1912 } else { 1913 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1914 } 1915} 1916 1917// ---------------------------------------------------------------------------- 1918 1919AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1920 audio_io_handle_t id, uint32_t device, type_t type) 1921 : PlaybackThread(audioFlinger, output, id, device, type), 1922 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1923 mPrevMixerStatus(MIXER_IDLE) 1924{ 1925 // FIXME - Current mixer implementation only supports stereo output 1926 if (mChannelCount == 1) { 1927 ALOGE("Invalid audio hardware channel count"); 1928 } 1929} 1930 1931AudioFlinger::MixerThread::~MixerThread() 1932{ 1933 delete mAudioMixer; 1934} 1935 1936class CpuStats { 1937public: 1938 void sample(); 1939#ifdef DEBUG_CPU_USAGE 1940private: 1941 ThreadCpuUsage mCpu; 1942#endif 1943}; 1944 1945void CpuStats::sample() { 1946#ifdef DEBUG_CPU_USAGE 1947 const CentralTendencyStatistics& stats = mCpu.statistics(); 1948 mCpu.sampleAndEnable(); 1949 unsigned n = stats.n(); 1950 // mCpu.elapsed() is expensive, so don't call it every loop 1951 if ((n & 127) == 1) { 1952 long long elapsed = mCpu.elapsed(); 1953 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1954 double perLoop = elapsed / (double) n; 1955 double perLoop100 = perLoop * 0.01; 1956 double mean = stats.mean(); 1957 double stddev = stats.stddev(); 1958 double minimum = stats.minimum(); 1959 double maximum = stats.maximum(); 1960 mCpu.resetStatistics(); 1961 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1962 elapsed * .000000001, n, perLoop * .000001, 1963 mean * .001, 1964 stddev * .001, 1965 minimum * .001, 1966 maximum * .001, 1967 mean / perLoop100, 1968 stddev / perLoop100, 1969 minimum / perLoop100, 1970 maximum / perLoop100); 1971 } 1972 } 1973#endif 1974}; 1975 1976void AudioFlinger::PlaybackThread::checkSilentMode_l() 1977{ 1978 if (!mMasterMute) { 1979 char value[PROPERTY_VALUE_MAX]; 1980 if (property_get("ro.audio.silent", value, "0") > 0) { 1981 char *endptr; 1982 unsigned long ul = strtoul(value, &endptr, 0); 1983 if (*endptr == '\0' && ul != 0) { 1984 ALOGD("Silence is golden"); 1985 // The setprop command will not allow a property to be changed after 1986 // the first time it is set, so we don't have to worry about un-muting. 1987 setMasterMute_l(true); 1988 } 1989 } 1990 } 1991} 1992 1993bool AudioFlinger::MixerThread::threadLoop() 1994{ 1995 Vector< sp<Track> > tracksToRemove; 1996 nsecs_t standbyTime = systemTime(); 1997 size_t mixBufferSize = mFrameCount * mFrameSize; 1998 // FIXME: Relaxed timing because of a certain device that can't meet latency 1999 // Should be reduced to 2x after the vendor fixes the driver issue 2000 // increase threshold again due to low power audio mode. The way this warning threshold is 2001 // calculated and its usefulness should be reconsidered anyway. 2002 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2003 nsecs_t lastWarning = 0; 2004 bool longStandbyExit = false; 2005 uint32_t activeSleepTime = activeSleepTimeUs(); 2006 uint32_t idleSleepTime = idleSleepTimeUs(); 2007 uint32_t sleepTime = idleSleepTime; 2008 uint32_t sleepTimeShift = 0; 2009 Vector< sp<EffectChain> > effectChains; 2010 CpuStats cpuStats; 2011 2012 acquireWakeLock(); 2013 2014 while (!exitPending()) 2015 { 2016 cpuStats.sample(); 2017 processConfigEvents(); 2018 2019 mixer_state mixerStatus = MIXER_IDLE; 2020 { // scope for mLock 2021 2022 Mutex::Autolock _l(mLock); 2023 2024 if (checkForNewParameters_l()) { 2025 mixBufferSize = mFrameCount * mFrameSize; 2026 // FIXME: Relaxed timing because of a certain device that can't meet latency 2027 // Should be reduced to 2x after the vendor fixes the driver issue 2028 // increase threshold again due to low power audio mode. The way this warning 2029 // threshold is calculated and its usefulness should be reconsidered anyway. 2030 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2031 activeSleepTime = activeSleepTimeUs(); 2032 idleSleepTime = idleSleepTimeUs(); 2033 } 2034 2035 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2036 2037 // put audio hardware into standby after short delay 2038 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2039 mSuspended)) { 2040 if (!mStandby) { 2041 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2042 mOutput->stream->common.standby(&mOutput->stream->common); 2043 mStandby = true; 2044 mBytesWritten = 0; 2045 } 2046 2047 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2048 // we're about to wait, flush the binder command buffer 2049 IPCThreadState::self()->flushCommands(); 2050 2051 if (exitPending()) break; 2052 2053 releaseWakeLock_l(); 2054 // wait until we have something to do... 2055 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 2056 mWaitWorkCV.wait(mLock); 2057 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 2058 acquireWakeLock_l(); 2059 2060 mPrevMixerStatus = MIXER_IDLE; 2061 checkSilentMode_l(); 2062 2063 standbyTime = systemTime() + mStandbyTimeInNsecs; 2064 sleepTime = idleSleepTime; 2065 sleepTimeShift = 0; 2066 continue; 2067 } 2068 } 2069 2070 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2071 2072 // prevent any changes in effect chain list and in each effect chain 2073 // during mixing and effect process as the audio buffers could be deleted 2074 // or modified if an effect is created or deleted 2075 lockEffectChains_l(effectChains); 2076 } 2077 2078 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2079 // obtain the presentation timestamp of the next output buffer 2080 int64_t pts; 2081 status_t status = INVALID_OPERATION; 2082 2083 if (NULL != mOutput->stream->get_next_write_timestamp) { 2084 status = mOutput->stream->get_next_write_timestamp( 2085 mOutput->stream, &pts); 2086 } 2087 2088 if (status != NO_ERROR) { 2089 pts = AudioBufferProvider::kInvalidPTS; 2090 } 2091 2092 // mix buffers... 2093 mAudioMixer->process(pts); 2094 // increase sleep time progressively when application underrun condition clears. 2095 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2096 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2097 // such that we would underrun the audio HAL. 2098 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2099 sleepTimeShift--; 2100 } 2101 sleepTime = 0; 2102 standbyTime = systemTime() + mStandbyTimeInNsecs; 2103 //TODO: delay standby when effects have a tail 2104 } else { 2105 // If no tracks are ready, sleep once for the duration of an output 2106 // buffer size, then write 0s to the output 2107 if (sleepTime == 0) { 2108 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2109 sleepTime = activeSleepTime >> sleepTimeShift; 2110 if (sleepTime < kMinThreadSleepTimeUs) { 2111 sleepTime = kMinThreadSleepTimeUs; 2112 } 2113 // reduce sleep time in case of consecutive application underruns to avoid 2114 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2115 // duration we would end up writing less data than needed by the audio HAL if 2116 // the condition persists. 2117 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2118 sleepTimeShift++; 2119 } 2120 } else { 2121 sleepTime = idleSleepTime; 2122 } 2123 } else if (mBytesWritten != 0 || 2124 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2125 memset (mMixBuffer, 0, mixBufferSize); 2126 sleepTime = 0; 2127 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2128 } 2129 // TODO add standby time extension fct of effect tail 2130 } 2131 2132 if (mSuspended) { 2133 sleepTime = suspendSleepTimeUs(); 2134 } 2135 // sleepTime == 0 means we must write to audio hardware 2136 if (sleepTime == 0) { 2137 for (size_t i = 0; i < effectChains.size(); i ++) { 2138 effectChains[i]->process_l(); 2139 } 2140 // enable changes in effect chain 2141 unlockEffectChains(effectChains); 2142 mLastWriteTime = systemTime(); 2143 mInWrite = true; 2144 mBytesWritten += mixBufferSize; 2145 2146 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2147 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2148 mNumWrites++; 2149 mInWrite = false; 2150 nsecs_t now = systemTime(); 2151 nsecs_t delta = now - mLastWriteTime; 2152 if (!mStandby && delta > maxPeriod) { 2153 mNumDelayedWrites++; 2154 if ((now - lastWarning) > kWarningThrottleNs) { 2155 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2156 ns2ms(delta), mNumDelayedWrites, this); 2157 lastWarning = now; 2158 } 2159 if (mStandby) { 2160 longStandbyExit = true; 2161 } 2162 } 2163 mStandby = false; 2164 } else { 2165 // enable changes in effect chain 2166 unlockEffectChains(effectChains); 2167 usleep(sleepTime); 2168 } 2169 2170 // finally let go of all our tracks, without the lock held 2171 // since we can't guarantee the destructors won't acquire that 2172 // same lock. 2173 tracksToRemove.clear(); 2174 2175 // Effect chains will be actually deleted here if they were removed from 2176 // mEffectChains list during mixing or effects processing 2177 effectChains.clear(); 2178 } 2179 2180 if (!mStandby) { 2181 mOutput->stream->common.standby(&mOutput->stream->common); 2182 } 2183 2184 releaseWakeLock(); 2185 2186 ALOGV("MixerThread %p exiting", this); 2187 return false; 2188} 2189 2190// prepareTracks_l() must be called with ThreadBase::mLock held 2191AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2192 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2193{ 2194 2195 mixer_state mixerStatus = MIXER_IDLE; 2196 // find out which tracks need to be processed 2197 size_t count = activeTracks.size(); 2198 size_t mixedTracks = 0; 2199 size_t tracksWithEffect = 0; 2200 2201 float masterVolume = mMasterVolume; 2202 bool masterMute = mMasterMute; 2203 2204 if (masterMute) { 2205 masterVolume = 0; 2206 } 2207 // Delegate master volume control to effect in output mix effect chain if needed 2208 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2209 if (chain != 0) { 2210 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2211 chain->setVolume_l(&v, &v); 2212 masterVolume = (float)((v + (1 << 23)) >> 24); 2213 chain.clear(); 2214 } 2215 2216 for (size_t i=0 ; i<count ; i++) { 2217 sp<Track> t = activeTracks[i].promote(); 2218 if (t == 0) continue; 2219 2220 // this const just means the local variable doesn't change 2221 Track* const track = t.get(); 2222 audio_track_cblk_t* cblk = track->cblk(); 2223 2224 // The first time a track is added we wait 2225 // for all its buffers to be filled before processing it 2226 int name = track->name(); 2227 // make sure that we have enough frames to mix one full buffer. 2228 // enforce this condition only once to enable draining the buffer in case the client 2229 // app does not call stop() and relies on underrun to stop: 2230 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2231 // during last round 2232 uint32_t minFrames = 1; 2233 if (!track->isStopped() && !track->isPausing() && 2234 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2235 if (t->sampleRate() == (int)mSampleRate) { 2236 minFrames = mFrameCount; 2237 } else { 2238 // +1 for rounding and +1 for additional sample needed for interpolation 2239 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2240 // add frames already consumed but not yet released by the resampler 2241 // because cblk->framesReady() will include these frames 2242 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2243 // the minimum track buffer size is normally twice the number of frames necessary 2244 // to fill one buffer and the resampler should not leave more than one buffer worth 2245 // of unreleased frames after each pass, but just in case... 2246 ALOG_ASSERT(minFrames <= cblk->frameCount); 2247 } 2248 } 2249 if ((track->framesReady() >= minFrames) && track->isReady() && 2250 !track->isPaused() && !track->isTerminated()) 2251 { 2252 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2253 2254 mixedTracks++; 2255 2256 // track->mainBuffer() != mMixBuffer means there is an effect chain 2257 // connected to the track 2258 chain.clear(); 2259 if (track->mainBuffer() != mMixBuffer) { 2260 chain = getEffectChain_l(track->sessionId()); 2261 // Delegate volume control to effect in track effect chain if needed 2262 if (chain != 0) { 2263 tracksWithEffect++; 2264 } else { 2265 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2266 name, track->sessionId()); 2267 } 2268 } 2269 2270 2271 int param = AudioMixer::VOLUME; 2272 if (track->mFillingUpStatus == Track::FS_FILLED) { 2273 // no ramp for the first volume setting 2274 track->mFillingUpStatus = Track::FS_ACTIVE; 2275 if (track->mState == TrackBase::RESUMING) { 2276 track->mState = TrackBase::ACTIVE; 2277 param = AudioMixer::RAMP_VOLUME; 2278 } 2279 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2280 } else if (cblk->server != 0) { 2281 // If the track is stopped before the first frame was mixed, 2282 // do not apply ramp 2283 param = AudioMixer::RAMP_VOLUME; 2284 } 2285 2286 // compute volume for this track 2287 uint32_t vl, vr, va; 2288 if (track->isMuted() || track->isPausing() || 2289 mStreamTypes[track->streamType()].mute) { 2290 vl = vr = va = 0; 2291 if (track->isPausing()) { 2292 track->setPaused(); 2293 } 2294 } else { 2295 2296 // read original volumes with volume control 2297 float typeVolume = mStreamTypes[track->streamType()].volume; 2298 float v = masterVolume * typeVolume; 2299 uint32_t vlr = cblk->getVolumeLR(); 2300 vl = vlr & 0xFFFF; 2301 vr = vlr >> 16; 2302 // track volumes come from shared memory, so can't be trusted and must be clamped 2303 if (vl > MAX_GAIN_INT) { 2304 ALOGV("Track left volume out of range: %04X", vl); 2305 vl = MAX_GAIN_INT; 2306 } 2307 if (vr > MAX_GAIN_INT) { 2308 ALOGV("Track right volume out of range: %04X", vr); 2309 vr = MAX_GAIN_INT; 2310 } 2311 // now apply the master volume and stream type volume 2312 vl = (uint32_t)(v * vl) << 12; 2313 vr = (uint32_t)(v * vr) << 12; 2314 // assuming master volume and stream type volume each go up to 1.0, 2315 // vl and vr are now in 8.24 format 2316 2317 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2318 // send level comes from shared memory and so may be corrupt 2319 if (sendLevel > MAX_GAIN_INT) { 2320 ALOGV("Track send level out of range: %04X", sendLevel); 2321 sendLevel = MAX_GAIN_INT; 2322 } 2323 va = (uint32_t)(v * sendLevel); 2324 } 2325 // Delegate volume control to effect in track effect chain if needed 2326 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2327 // Do not ramp volume if volume is controlled by effect 2328 param = AudioMixer::VOLUME; 2329 track->mHasVolumeController = true; 2330 } else { 2331 // force no volume ramp when volume controller was just disabled or removed 2332 // from effect chain to avoid volume spike 2333 if (track->mHasVolumeController) { 2334 param = AudioMixer::VOLUME; 2335 } 2336 track->mHasVolumeController = false; 2337 } 2338 2339 // Convert volumes from 8.24 to 4.12 format 2340 // This additional clamping is needed in case chain->setVolume_l() overshot 2341 vl = (vl + (1 << 11)) >> 12; 2342 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2343 vr = (vr + (1 << 11)) >> 12; 2344 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2345 2346 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2347 2348 // XXX: these things DON'T need to be done each time 2349 mAudioMixer->setBufferProvider(name, track); 2350 mAudioMixer->enable(name); 2351 2352 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2353 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2354 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2355 mAudioMixer->setParameter( 2356 name, 2357 AudioMixer::TRACK, 2358 AudioMixer::FORMAT, (void *)track->format()); 2359 mAudioMixer->setParameter( 2360 name, 2361 AudioMixer::TRACK, 2362 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2363 mAudioMixer->setParameter( 2364 name, 2365 AudioMixer::RESAMPLE, 2366 AudioMixer::SAMPLE_RATE, 2367 (void *)(cblk->sampleRate)); 2368 mAudioMixer->setParameter( 2369 name, 2370 AudioMixer::TRACK, 2371 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2372 mAudioMixer->setParameter( 2373 name, 2374 AudioMixer::TRACK, 2375 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2376 2377 // reset retry count 2378 track->mRetryCount = kMaxTrackRetries; 2379 // If one track is ready, set the mixer ready if: 2380 // - the mixer was not ready during previous round OR 2381 // - no other track is not ready 2382 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2383 mixerStatus != MIXER_TRACKS_ENABLED) { 2384 mixerStatus = MIXER_TRACKS_READY; 2385 } 2386 } else { 2387 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2388 if (track->isStopped()) { 2389 track->reset(); 2390 } 2391 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2392 // We have consumed all the buffers of this track. 2393 // Remove it from the list of active tracks. 2394 tracksToRemove->add(track); 2395 } else { 2396 // No buffers for this track. Give it a few chances to 2397 // fill a buffer, then remove it from active list. 2398 if (--(track->mRetryCount) <= 0) { 2399 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2400 tracksToRemove->add(track); 2401 // indicate to client process that the track was disabled because of underrun 2402 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2403 // If one track is not ready, mark the mixer also not ready if: 2404 // - the mixer was ready during previous round OR 2405 // - no other track is ready 2406 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2407 mixerStatus != MIXER_TRACKS_READY) { 2408 mixerStatus = MIXER_TRACKS_ENABLED; 2409 } 2410 } 2411 mAudioMixer->disable(name); 2412 } 2413 } 2414 2415 // remove all the tracks that need to be... 2416 count = tracksToRemove->size(); 2417 if (CC_UNLIKELY(count)) { 2418 for (size_t i=0 ; i<count ; i++) { 2419 const sp<Track>& track = tracksToRemove->itemAt(i); 2420 mActiveTracks.remove(track); 2421 if (track->mainBuffer() != mMixBuffer) { 2422 chain = getEffectChain_l(track->sessionId()); 2423 if (chain != 0) { 2424 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2425 chain->decActiveTrackCnt(); 2426 } 2427 } 2428 if (track->isTerminated()) { 2429 removeTrack_l(track); 2430 } 2431 } 2432 } 2433 2434 // mix buffer must be cleared if all tracks are connected to an 2435 // effect chain as in this case the mixer will not write to 2436 // mix buffer and track effects will accumulate into it 2437 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2438 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2439 } 2440 2441 mPrevMixerStatus = mixerStatus; 2442 return mixerStatus; 2443} 2444 2445void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2446{ 2447 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2448 this, streamType, mTracks.size()); 2449 Mutex::Autolock _l(mLock); 2450 2451 size_t size = mTracks.size(); 2452 for (size_t i = 0; i < size; i++) { 2453 sp<Track> t = mTracks[i]; 2454 if (t->streamType() == streamType) { 2455 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2456 t->mCblk->cv.signal(); 2457 } 2458 } 2459} 2460 2461void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2462{ 2463 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2464 this, streamType, valid); 2465 Mutex::Autolock _l(mLock); 2466 2467 mStreamTypes[streamType].valid = valid; 2468} 2469 2470// getTrackName_l() must be called with ThreadBase::mLock held 2471int AudioFlinger::MixerThread::getTrackName_l() 2472{ 2473 return mAudioMixer->getTrackName(); 2474} 2475 2476// deleteTrackName_l() must be called with ThreadBase::mLock held 2477void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2478{ 2479 ALOGV("remove track (%d) and delete from mixer", name); 2480 mAudioMixer->deleteTrackName(name); 2481} 2482 2483// checkForNewParameters_l() must be called with ThreadBase::mLock held 2484bool AudioFlinger::MixerThread::checkForNewParameters_l() 2485{ 2486 bool reconfig = false; 2487 2488 while (!mNewParameters.isEmpty()) { 2489 status_t status = NO_ERROR; 2490 String8 keyValuePair = mNewParameters[0]; 2491 AudioParameter param = AudioParameter(keyValuePair); 2492 int value; 2493 2494 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2495 reconfig = true; 2496 } 2497 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2498 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2499 status = BAD_VALUE; 2500 } else { 2501 reconfig = true; 2502 } 2503 } 2504 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2505 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2506 status = BAD_VALUE; 2507 } else { 2508 reconfig = true; 2509 } 2510 } 2511 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2512 // do not accept frame count changes if tracks are open as the track buffer 2513 // size depends on frame count and correct behavior would not be guaranteed 2514 // if frame count is changed after track creation 2515 if (!mTracks.isEmpty()) { 2516 status = INVALID_OPERATION; 2517 } else { 2518 reconfig = true; 2519 } 2520 } 2521 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2522 // when changing the audio output device, call addBatteryData to notify 2523 // the change 2524 if ((int)mDevice != value) { 2525 uint32_t params = 0; 2526 // check whether speaker is on 2527 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2528 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2529 } 2530 2531 int deviceWithoutSpeaker 2532 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2533 // check if any other device (except speaker) is on 2534 if (value & deviceWithoutSpeaker ) { 2535 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2536 } 2537 2538 if (params != 0) { 2539 addBatteryData(params); 2540 } 2541 } 2542 2543 // forward device change to effects that have requested to be 2544 // aware of attached audio device. 2545 mDevice = (uint32_t)value; 2546 for (size_t i = 0; i < mEffectChains.size(); i++) { 2547 mEffectChains[i]->setDevice_l(mDevice); 2548 } 2549 } 2550 2551 if (status == NO_ERROR) { 2552 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2553 keyValuePair.string()); 2554 if (!mStandby && status == INVALID_OPERATION) { 2555 mOutput->stream->common.standby(&mOutput->stream->common); 2556 mStandby = true; 2557 mBytesWritten = 0; 2558 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2559 keyValuePair.string()); 2560 } 2561 if (status == NO_ERROR && reconfig) { 2562 delete mAudioMixer; 2563 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2564 mAudioMixer = NULL; 2565 readOutputParameters(); 2566 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2567 for (size_t i = 0; i < mTracks.size() ; i++) { 2568 int name = getTrackName_l(); 2569 if (name < 0) break; 2570 mTracks[i]->mName = name; 2571 // limit track sample rate to 2 x new output sample rate 2572 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2573 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2574 } 2575 } 2576 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2577 } 2578 } 2579 2580 mNewParameters.removeAt(0); 2581 2582 mParamStatus = status; 2583 mParamCond.signal(); 2584 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2585 // already timed out waiting for the status and will never signal the condition. 2586 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2587 } 2588 return reconfig; 2589} 2590 2591status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2592{ 2593 const size_t SIZE = 256; 2594 char buffer[SIZE]; 2595 String8 result; 2596 2597 PlaybackThread::dumpInternals(fd, args); 2598 2599 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2600 result.append(buffer); 2601 write(fd, result.string(), result.size()); 2602 return NO_ERROR; 2603} 2604 2605uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2606{ 2607 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2608} 2609 2610uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2611{ 2612 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2613} 2614 2615// ---------------------------------------------------------------------------- 2616AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2617 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2618 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2619 // mLeftVolFloat, mRightVolFloat 2620 // mLeftVolShort, mRightVolShort 2621{ 2622} 2623 2624AudioFlinger::DirectOutputThread::~DirectOutputThread() 2625{ 2626} 2627 2628void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2629{ 2630 // Do not apply volume on compressed audio 2631 if (!audio_is_linear_pcm(mFormat)) { 2632 return; 2633 } 2634 2635 // convert to signed 16 bit before volume calculation 2636 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2637 size_t count = mFrameCount * mChannelCount; 2638 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2639 int16_t *dst = mMixBuffer + count-1; 2640 while(count--) { 2641 *dst-- = (int16_t)(*src--^0x80) << 8; 2642 } 2643 } 2644 2645 size_t frameCount = mFrameCount; 2646 int16_t *out = mMixBuffer; 2647 if (ramp) { 2648 if (mChannelCount == 1) { 2649 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2650 int32_t vlInc = d / (int32_t)frameCount; 2651 int32_t vl = ((int32_t)mLeftVolShort << 16); 2652 do { 2653 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2654 out++; 2655 vl += vlInc; 2656 } while (--frameCount); 2657 2658 } else { 2659 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2660 int32_t vlInc = d / (int32_t)frameCount; 2661 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2662 int32_t vrInc = d / (int32_t)frameCount; 2663 int32_t vl = ((int32_t)mLeftVolShort << 16); 2664 int32_t vr = ((int32_t)mRightVolShort << 16); 2665 do { 2666 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2667 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2668 out += 2; 2669 vl += vlInc; 2670 vr += vrInc; 2671 } while (--frameCount); 2672 } 2673 } else { 2674 if (mChannelCount == 1) { 2675 do { 2676 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2677 out++; 2678 } while (--frameCount); 2679 } else { 2680 do { 2681 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2682 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2683 out += 2; 2684 } while (--frameCount); 2685 } 2686 } 2687 2688 // convert back to unsigned 8 bit after volume calculation 2689 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2690 size_t count = mFrameCount * mChannelCount; 2691 int16_t *src = mMixBuffer; 2692 uint8_t *dst = (uint8_t *)mMixBuffer; 2693 while(count--) { 2694 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2695 } 2696 } 2697 2698 mLeftVolShort = leftVol; 2699 mRightVolShort = rightVol; 2700} 2701 2702bool AudioFlinger::DirectOutputThread::threadLoop() 2703{ 2704 sp<Track> trackToRemove; 2705 sp<Track> activeTrack; 2706 nsecs_t standbyTime = systemTime(); 2707 size_t mixBufferSize = mFrameCount*mFrameSize; 2708 uint32_t activeSleepTime = activeSleepTimeUs(); 2709 uint32_t idleSleepTime = idleSleepTimeUs(); 2710 uint32_t sleepTime = idleSleepTime; 2711 // use shorter standby delay as on normal output to release 2712 // hardware resources as soon as possible 2713 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2714 2715 acquireWakeLock(); 2716 2717 while (!exitPending()) 2718 { 2719 bool rampVolume; 2720 uint16_t leftVol; 2721 uint16_t rightVol; 2722 Vector< sp<EffectChain> > effectChains; 2723 2724 processConfigEvents(); 2725 2726 mixer_state mixerStatus = MIXER_IDLE; 2727 { // scope for the mLock 2728 2729 Mutex::Autolock _l(mLock); 2730 2731 if (checkForNewParameters_l()) { 2732 mixBufferSize = mFrameCount*mFrameSize; 2733 activeSleepTime = activeSleepTimeUs(); 2734 idleSleepTime = idleSleepTimeUs(); 2735 standbyDelay = microseconds(activeSleepTime*2); 2736 } 2737 2738 // put audio hardware into standby after short delay 2739 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2740 mSuspended)) { 2741 // wait until we have something to do... 2742 if (!mStandby) { 2743 ALOGV("Audio hardware entering standby, mixer %p", this); 2744 mOutput->stream->common.standby(&mOutput->stream->common); 2745 mStandby = true; 2746 mBytesWritten = 0; 2747 } 2748 2749 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2750 // we're about to wait, flush the binder command buffer 2751 IPCThreadState::self()->flushCommands(); 2752 2753 if (exitPending()) break; 2754 2755 releaseWakeLock_l(); 2756 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2757 mWaitWorkCV.wait(mLock); 2758 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2759 acquireWakeLock_l(); 2760 2761 checkSilentMode_l(); 2762 2763 standbyTime = systemTime() + standbyDelay; 2764 sleepTime = idleSleepTime; 2765 continue; 2766 } 2767 } 2768 2769 effectChains = mEffectChains; 2770 2771 // find out which tracks need to be processed 2772 if (mActiveTracks.size() != 0) { 2773 sp<Track> t = mActiveTracks[0].promote(); 2774 if (t == 0) continue; 2775 2776 Track* const track = t.get(); 2777 audio_track_cblk_t* cblk = track->cblk(); 2778 2779 // The first time a track is added we wait 2780 // for all its buffers to be filled before processing it 2781 if (cblk->framesReady() && track->isReady() && 2782 !track->isPaused() && !track->isTerminated()) 2783 { 2784 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2785 2786 if (track->mFillingUpStatus == Track::FS_FILLED) { 2787 track->mFillingUpStatus = Track::FS_ACTIVE; 2788 mLeftVolFloat = mRightVolFloat = 0; 2789 mLeftVolShort = mRightVolShort = 0; 2790 if (track->mState == TrackBase::RESUMING) { 2791 track->mState = TrackBase::ACTIVE; 2792 rampVolume = true; 2793 } 2794 } else if (cblk->server != 0) { 2795 // If the track is stopped before the first frame was mixed, 2796 // do not apply ramp 2797 rampVolume = true; 2798 } 2799 // compute volume for this track 2800 float left, right; 2801 if (track->isMuted() || mMasterMute || track->isPausing() || 2802 mStreamTypes[track->streamType()].mute) { 2803 left = right = 0; 2804 if (track->isPausing()) { 2805 track->setPaused(); 2806 } 2807 } else { 2808 float typeVolume = mStreamTypes[track->streamType()].volume; 2809 float v = mMasterVolume * typeVolume; 2810 uint32_t vlr = cblk->getVolumeLR(); 2811 float v_clamped = v * (vlr & 0xFFFF); 2812 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2813 left = v_clamped/MAX_GAIN; 2814 v_clamped = v * (vlr >> 16); 2815 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2816 right = v_clamped/MAX_GAIN; 2817 } 2818 2819 if (left != mLeftVolFloat || right != mRightVolFloat) { 2820 mLeftVolFloat = left; 2821 mRightVolFloat = right; 2822 2823 // If audio HAL implements volume control, 2824 // force software volume to nominal value 2825 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2826 left = 1.0f; 2827 right = 1.0f; 2828 } 2829 2830 // Convert volumes from float to 8.24 2831 uint32_t vl = (uint32_t)(left * (1 << 24)); 2832 uint32_t vr = (uint32_t)(right * (1 << 24)); 2833 2834 // Delegate volume control to effect in track effect chain if needed 2835 // only one effect chain can be present on DirectOutputThread, so if 2836 // there is one, the track is connected to it 2837 if (!effectChains.isEmpty()) { 2838 // Do not ramp volume if volume is controlled by effect 2839 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2840 rampVolume = false; 2841 } 2842 } 2843 2844 // Convert volumes from 8.24 to 4.12 format 2845 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2846 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2847 leftVol = (uint16_t)v_clamped; 2848 v_clamped = (vr + (1 << 11)) >> 12; 2849 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2850 rightVol = (uint16_t)v_clamped; 2851 } else { 2852 leftVol = mLeftVolShort; 2853 rightVol = mRightVolShort; 2854 rampVolume = false; 2855 } 2856 2857 // reset retry count 2858 track->mRetryCount = kMaxTrackRetriesDirect; 2859 activeTrack = t; 2860 mixerStatus = MIXER_TRACKS_READY; 2861 } else { 2862 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2863 if (track->isStopped()) { 2864 track->reset(); 2865 } 2866 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2867 // We have consumed all the buffers of this track. 2868 // Remove it from the list of active tracks. 2869 trackToRemove = track; 2870 } else { 2871 // No buffers for this track. Give it a few chances to 2872 // fill a buffer, then remove it from active list. 2873 if (--(track->mRetryCount) <= 0) { 2874 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2875 trackToRemove = track; 2876 } else { 2877 mixerStatus = MIXER_TRACKS_ENABLED; 2878 } 2879 } 2880 } 2881 } 2882 2883 // remove all the tracks that need to be... 2884 if (CC_UNLIKELY(trackToRemove != 0)) { 2885 mActiveTracks.remove(trackToRemove); 2886 if (!effectChains.isEmpty()) { 2887 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2888 trackToRemove->sessionId()); 2889 effectChains[0]->decActiveTrackCnt(); 2890 } 2891 if (trackToRemove->isTerminated()) { 2892 removeTrack_l(trackToRemove); 2893 } 2894 } 2895 2896 lockEffectChains_l(effectChains); 2897 } 2898 2899 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2900 AudioBufferProvider::Buffer buffer; 2901 size_t frameCount = mFrameCount; 2902 int8_t *curBuf = (int8_t *)mMixBuffer; 2903 // output audio to hardware 2904 while (frameCount) { 2905 buffer.frameCount = frameCount; 2906 activeTrack->getNextBuffer(&buffer, 2907 AudioBufferProvider::kInvalidPTS); 2908 if (CC_UNLIKELY(buffer.raw == NULL)) { 2909 memset(curBuf, 0, frameCount * mFrameSize); 2910 break; 2911 } 2912 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2913 frameCount -= buffer.frameCount; 2914 curBuf += buffer.frameCount * mFrameSize; 2915 activeTrack->releaseBuffer(&buffer); 2916 } 2917 sleepTime = 0; 2918 standbyTime = systemTime() + standbyDelay; 2919 } else { 2920 if (sleepTime == 0) { 2921 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2922 sleepTime = activeSleepTime; 2923 } else { 2924 sleepTime = idleSleepTime; 2925 } 2926 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2927 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2928 sleepTime = 0; 2929 } 2930 } 2931 2932 if (mSuspended) { 2933 sleepTime = suspendSleepTimeUs(); 2934 } 2935 // sleepTime == 0 means we must write to audio hardware 2936 if (sleepTime == 0) { 2937 if (mixerStatus == MIXER_TRACKS_READY) { 2938 applyVolume(leftVol, rightVol, rampVolume); 2939 } 2940 for (size_t i = 0; i < effectChains.size(); i ++) { 2941 effectChains[i]->process_l(); 2942 } 2943 unlockEffectChains(effectChains); 2944 2945 mLastWriteTime = systemTime(); 2946 mInWrite = true; 2947 mBytesWritten += mixBufferSize; 2948 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2949 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2950 mNumWrites++; 2951 mInWrite = false; 2952 mStandby = false; 2953 } else { 2954 unlockEffectChains(effectChains); 2955 usleep(sleepTime); 2956 } 2957 2958 // finally let go of removed track, without the lock held 2959 // since we can't guarantee the destructors won't acquire that 2960 // same lock. 2961 trackToRemove.clear(); 2962 activeTrack.clear(); 2963 2964 // Effect chains will be actually deleted here if they were removed from 2965 // mEffectChains list during mixing or effects processing 2966 effectChains.clear(); 2967 } 2968 2969 if (!mStandby) { 2970 mOutput->stream->common.standby(&mOutput->stream->common); 2971 } 2972 2973 releaseWakeLock(); 2974 2975 ALOGV("DirectOutputThread %p exiting", this); 2976 return false; 2977} 2978 2979// getTrackName_l() must be called with ThreadBase::mLock held 2980int AudioFlinger::DirectOutputThread::getTrackName_l() 2981{ 2982 return 0; 2983} 2984 2985// deleteTrackName_l() must be called with ThreadBase::mLock held 2986void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2987{ 2988} 2989 2990// checkForNewParameters_l() must be called with ThreadBase::mLock held 2991bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2992{ 2993 bool reconfig = false; 2994 2995 while (!mNewParameters.isEmpty()) { 2996 status_t status = NO_ERROR; 2997 String8 keyValuePair = mNewParameters[0]; 2998 AudioParameter param = AudioParameter(keyValuePair); 2999 int value; 3000 3001 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3002 // do not accept frame count changes if tracks are open as the track buffer 3003 // size depends on frame count and correct behavior would not be garantied 3004 // if frame count is changed after track creation 3005 if (!mTracks.isEmpty()) { 3006 status = INVALID_OPERATION; 3007 } else { 3008 reconfig = true; 3009 } 3010 } 3011 if (status == NO_ERROR) { 3012 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3013 keyValuePair.string()); 3014 if (!mStandby && status == INVALID_OPERATION) { 3015 mOutput->stream->common.standby(&mOutput->stream->common); 3016 mStandby = true; 3017 mBytesWritten = 0; 3018 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3019 keyValuePair.string()); 3020 } 3021 if (status == NO_ERROR && reconfig) { 3022 readOutputParameters(); 3023 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3024 } 3025 } 3026 3027 mNewParameters.removeAt(0); 3028 3029 mParamStatus = status; 3030 mParamCond.signal(); 3031 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3032 // already timed out waiting for the status and will never signal the condition. 3033 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3034 } 3035 return reconfig; 3036} 3037 3038uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3039{ 3040 uint32_t time; 3041 if (audio_is_linear_pcm(mFormat)) { 3042 time = PlaybackThread::activeSleepTimeUs(); 3043 } else { 3044 time = 10000; 3045 } 3046 return time; 3047} 3048 3049uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3050{ 3051 uint32_t time; 3052 if (audio_is_linear_pcm(mFormat)) { 3053 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3054 } else { 3055 time = 10000; 3056 } 3057 return time; 3058} 3059 3060uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3061{ 3062 uint32_t time; 3063 if (audio_is_linear_pcm(mFormat)) { 3064 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3065 } else { 3066 time = 10000; 3067 } 3068 return time; 3069} 3070 3071 3072// ---------------------------------------------------------------------------- 3073 3074AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3075 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3076 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3077 mWaitTimeMs(UINT_MAX) 3078{ 3079 addOutputTrack(mainThread); 3080} 3081 3082AudioFlinger::DuplicatingThread::~DuplicatingThread() 3083{ 3084 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3085 mOutputTracks[i]->destroy(); 3086 } 3087} 3088 3089bool AudioFlinger::DuplicatingThread::threadLoop() 3090{ 3091 Vector< sp<Track> > tracksToRemove; 3092 nsecs_t standbyTime = systemTime(); 3093 size_t mixBufferSize = mFrameCount*mFrameSize; 3094 SortedVector< sp<OutputTrack> > outputTracks; 3095 uint32_t writeFrames = 0; 3096 uint32_t activeSleepTime = activeSleepTimeUs(); 3097 uint32_t idleSleepTime = idleSleepTimeUs(); 3098 uint32_t sleepTime = idleSleepTime; 3099 Vector< sp<EffectChain> > effectChains; 3100 3101 acquireWakeLock(); 3102 3103 while (!exitPending()) 3104 { 3105 processConfigEvents(); 3106 3107 mixer_state mixerStatus = MIXER_IDLE; 3108 { // scope for the mLock 3109 3110 Mutex::Autolock _l(mLock); 3111 3112 if (checkForNewParameters_l()) { 3113 mixBufferSize = mFrameCount*mFrameSize; 3114 updateWaitTime(); 3115 activeSleepTime = activeSleepTimeUs(); 3116 idleSleepTime = idleSleepTimeUs(); 3117 } 3118 3119 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3120 3121 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3122 outputTracks.add(mOutputTracks[i]); 3123 } 3124 3125 // put audio hardware into standby after short delay 3126 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3127 mSuspended)) { 3128 if (!mStandby) { 3129 for (size_t i = 0; i < outputTracks.size(); i++) { 3130 outputTracks[i]->stop(); 3131 } 3132 mStandby = true; 3133 mBytesWritten = 0; 3134 } 3135 3136 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3137 // we're about to wait, flush the binder command buffer 3138 IPCThreadState::self()->flushCommands(); 3139 outputTracks.clear(); 3140 3141 if (exitPending()) break; 3142 3143 releaseWakeLock_l(); 3144 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3145 mWaitWorkCV.wait(mLock); 3146 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3147 acquireWakeLock_l(); 3148 3149 checkSilentMode_l(); 3150 3151 standbyTime = systemTime() + mStandbyTimeInNsecs; 3152 sleepTime = idleSleepTime; 3153 continue; 3154 } 3155 } 3156 3157 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3158 3159 // prevent any changes in effect chain list and in each effect chain 3160 // during mixing and effect process as the audio buffers could be deleted 3161 // or modified if an effect is created or deleted 3162 lockEffectChains_l(effectChains); 3163 } 3164 3165 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3166 // mix buffers... 3167 if (outputsReady(outputTracks)) { 3168 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3169 } else { 3170 memset(mMixBuffer, 0, mixBufferSize); 3171 } 3172 sleepTime = 0; 3173 writeFrames = mFrameCount; 3174 } else { 3175 if (sleepTime == 0) { 3176 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3177 sleepTime = activeSleepTime; 3178 } else { 3179 sleepTime = idleSleepTime; 3180 } 3181 } else if (mBytesWritten != 0) { 3182 // flush remaining overflow buffers in output tracks 3183 for (size_t i = 0; i < outputTracks.size(); i++) { 3184 if (outputTracks[i]->isActive()) { 3185 sleepTime = 0; 3186 writeFrames = 0; 3187 memset(mMixBuffer, 0, mixBufferSize); 3188 break; 3189 } 3190 } 3191 } 3192 } 3193 3194 if (mSuspended) { 3195 sleepTime = suspendSleepTimeUs(); 3196 } 3197 // sleepTime == 0 means we must write to audio hardware 3198 if (sleepTime == 0) { 3199 for (size_t i = 0; i < effectChains.size(); i ++) { 3200 effectChains[i]->process_l(); 3201 } 3202 // enable changes in effect chain 3203 unlockEffectChains(effectChains); 3204 3205 standbyTime = systemTime() + mStandbyTimeInNsecs; 3206 for (size_t i = 0; i < outputTracks.size(); i++) { 3207 outputTracks[i]->write(mMixBuffer, writeFrames); 3208 } 3209 mStandby = false; 3210 mBytesWritten += mixBufferSize; 3211 } else { 3212 // enable changes in effect chain 3213 unlockEffectChains(effectChains); 3214 usleep(sleepTime); 3215 } 3216 3217 // finally let go of all our tracks, without the lock held 3218 // since we can't guarantee the destructors won't acquire that 3219 // same lock. 3220 tracksToRemove.clear(); 3221 outputTracks.clear(); 3222 3223 // Effect chains will be actually deleted here if they were removed from 3224 // mEffectChains list during mixing or effects processing 3225 effectChains.clear(); 3226 } 3227 3228 releaseWakeLock(); 3229 3230 return false; 3231} 3232 3233void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3234{ 3235 // FIXME explain this formula 3236 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3237 OutputTrack *outputTrack = new OutputTrack(thread, 3238 this, 3239 mSampleRate, 3240 mFormat, 3241 mChannelMask, 3242 frameCount); 3243 if (outputTrack->cblk() != NULL) { 3244 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3245 mOutputTracks.add(outputTrack); 3246 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3247 updateWaitTime(); 3248 } 3249} 3250 3251void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3252{ 3253 Mutex::Autolock _l(mLock); 3254 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3255 if (mOutputTracks[i]->thread() == thread) { 3256 mOutputTracks[i]->destroy(); 3257 mOutputTracks.removeAt(i); 3258 updateWaitTime(); 3259 return; 3260 } 3261 } 3262 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3263} 3264 3265void AudioFlinger::DuplicatingThread::updateWaitTime() 3266{ 3267 mWaitTimeMs = UINT_MAX; 3268 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3269 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3270 if (strong != 0) { 3271 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3272 if (waitTimeMs < mWaitTimeMs) { 3273 mWaitTimeMs = waitTimeMs; 3274 } 3275 } 3276 } 3277} 3278 3279 3280bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3281{ 3282 for (size_t i = 0; i < outputTracks.size(); i++) { 3283 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3284 if (thread == 0) { 3285 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3286 return false; 3287 } 3288 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3289 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3290 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3291 return false; 3292 } 3293 } 3294 return true; 3295} 3296 3297uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3298{ 3299 return (mWaitTimeMs * 1000) / 2; 3300} 3301 3302// ---------------------------------------------------------------------------- 3303 3304// TrackBase constructor must be called with AudioFlinger::mLock held 3305AudioFlinger::ThreadBase::TrackBase::TrackBase( 3306 ThreadBase *thread, 3307 const sp<Client>& client, 3308 uint32_t sampleRate, 3309 audio_format_t format, 3310 uint32_t channelMask, 3311 int frameCount, 3312 const sp<IMemory>& sharedBuffer, 3313 int sessionId) 3314 : RefBase(), 3315 mThread(thread), 3316 mClient(client), 3317 mCblk(NULL), 3318 // mBuffer 3319 // mBufferEnd 3320 mFrameCount(0), 3321 mState(IDLE), 3322 mFormat(format), 3323 mStepServerFailed(false), 3324 mSessionId(sessionId) 3325 // mChannelCount 3326 // mChannelMask 3327{ 3328 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3329 3330 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3331 size_t size = sizeof(audio_track_cblk_t); 3332 uint8_t channelCount = popcount(channelMask); 3333 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3334 if (sharedBuffer == 0) { 3335 size += bufferSize; 3336 } 3337 3338 if (client != NULL) { 3339 mCblkMemory = client->heap()->allocate(size); 3340 if (mCblkMemory != 0) { 3341 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3342 if (mCblk != NULL) { // construct the shared structure in-place. 3343 new(mCblk) audio_track_cblk_t(); 3344 // clear all buffers 3345 mCblk->frameCount = frameCount; 3346 mCblk->sampleRate = sampleRate; 3347 mChannelCount = channelCount; 3348 mChannelMask = channelMask; 3349 if (sharedBuffer == 0) { 3350 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3351 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3352 // Force underrun condition to avoid false underrun callback until first data is 3353 // written to buffer (other flags are cleared) 3354 mCblk->flags = CBLK_UNDERRUN_ON; 3355 } else { 3356 mBuffer = sharedBuffer->pointer(); 3357 } 3358 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3359 } 3360 } else { 3361 ALOGE("not enough memory for AudioTrack size=%u", size); 3362 client->heap()->dump("AudioTrack"); 3363 return; 3364 } 3365 } else { 3366 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3367 // construct the shared structure in-place. 3368 new(mCblk) audio_track_cblk_t(); 3369 // clear all buffers 3370 mCblk->frameCount = frameCount; 3371 mCblk->sampleRate = sampleRate; 3372 mChannelCount = channelCount; 3373 mChannelMask = channelMask; 3374 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3375 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3376 // Force underrun condition to avoid false underrun callback until first data is 3377 // written to buffer (other flags are cleared) 3378 mCblk->flags = CBLK_UNDERRUN_ON; 3379 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3380 } 3381} 3382 3383AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3384{ 3385 if (mCblk != NULL) { 3386 if (mClient == 0) { 3387 delete mCblk; 3388 } else { 3389 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3390 } 3391 } 3392 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3393 if (mClient != 0) { 3394 // Client destructor must run with AudioFlinger mutex locked 3395 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3396 // If the client's reference count drops to zero, the associated destructor 3397 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3398 // relying on the automatic clear() at end of scope. 3399 mClient.clear(); 3400 } 3401} 3402 3403void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3404{ 3405 buffer->raw = NULL; 3406 mFrameCount = buffer->frameCount; 3407 step(); 3408 buffer->frameCount = 0; 3409} 3410 3411bool AudioFlinger::ThreadBase::TrackBase::step() { 3412 bool result; 3413 audio_track_cblk_t* cblk = this->cblk(); 3414 3415 result = cblk->stepServer(mFrameCount); 3416 if (!result) { 3417 ALOGV("stepServer failed acquiring cblk mutex"); 3418 mStepServerFailed = true; 3419 } 3420 return result; 3421} 3422 3423void AudioFlinger::ThreadBase::TrackBase::reset() { 3424 audio_track_cblk_t* cblk = this->cblk(); 3425 3426 cblk->user = 0; 3427 cblk->server = 0; 3428 cblk->userBase = 0; 3429 cblk->serverBase = 0; 3430 mStepServerFailed = false; 3431 ALOGV("TrackBase::reset"); 3432} 3433 3434int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3435 return (int)mCblk->sampleRate; 3436} 3437 3438void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3439 audio_track_cblk_t* cblk = this->cblk(); 3440 size_t frameSize = cblk->frameSize; 3441 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3442 int8_t *bufferEnd = bufferStart + frames * frameSize; 3443 3444 // Check validity of returned pointer in case the track control block would have been corrupted. 3445 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3446 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3447 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3448 server %d, serverBase %d, user %d, userBase %d", 3449 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3450 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3451 return NULL; 3452 } 3453 3454 return bufferStart; 3455} 3456 3457// ---------------------------------------------------------------------------- 3458 3459// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3460AudioFlinger::PlaybackThread::Track::Track( 3461 PlaybackThread *thread, 3462 const sp<Client>& client, 3463 audio_stream_type_t streamType, 3464 uint32_t sampleRate, 3465 audio_format_t format, 3466 uint32_t channelMask, 3467 int frameCount, 3468 const sp<IMemory>& sharedBuffer, 3469 int sessionId) 3470 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3471 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3472 mAuxEffectId(0), mHasVolumeController(false) 3473{ 3474 if (mCblk != NULL) { 3475 if (thread != NULL) { 3476 mName = thread->getTrackName_l(); 3477 mMainBuffer = thread->mixBuffer(); 3478 } 3479 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3480 if (mName < 0) { 3481 ALOGE("no more track names available"); 3482 } 3483 mStreamType = streamType; 3484 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3485 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3486 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3487 } 3488} 3489 3490AudioFlinger::PlaybackThread::Track::~Track() 3491{ 3492 ALOGV("PlaybackThread::Track destructor"); 3493 sp<ThreadBase> thread = mThread.promote(); 3494 if (thread != 0) { 3495 Mutex::Autolock _l(thread->mLock); 3496 mState = TERMINATED; 3497 } 3498} 3499 3500void AudioFlinger::PlaybackThread::Track::destroy() 3501{ 3502 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3503 // by removing it from mTracks vector, so there is a risk that this Tracks's 3504 // destructor is called. As the destructor needs to lock mLock, 3505 // we must acquire a strong reference on this Track before locking mLock 3506 // here so that the destructor is called only when exiting this function. 3507 // On the other hand, as long as Track::destroy() is only called by 3508 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3509 // this Track with its member mTrack. 3510 sp<Track> keep(this); 3511 { // scope for mLock 3512 sp<ThreadBase> thread = mThread.promote(); 3513 if (thread != 0) { 3514 if (!isOutputTrack()) { 3515 if (mState == ACTIVE || mState == RESUMING) { 3516 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3517 3518 // to track the speaker usage 3519 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3520 } 3521 AudioSystem::releaseOutput(thread->id()); 3522 } 3523 Mutex::Autolock _l(thread->mLock); 3524 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3525 playbackThread->destroyTrack_l(this); 3526 } 3527 } 3528} 3529 3530void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3531{ 3532 uint32_t vlr = mCblk->getVolumeLR(); 3533 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", 3534 mName - AudioMixer::TRACK0, 3535 (mClient == 0) ? getpid_cached : mClient->pid(), 3536 mStreamType, 3537 mFormat, 3538 mChannelMask, 3539 mSessionId, 3540 mFrameCount, 3541 mState, 3542 mMute, 3543 mFillingUpStatus, 3544 mCblk->sampleRate, 3545 vlr & 0xFFFF, 3546 vlr >> 16, 3547 mCblk->server, 3548 mCblk->user, 3549 (int)mMainBuffer, 3550 (int)mAuxBuffer); 3551} 3552 3553status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3554 AudioBufferProvider::Buffer* buffer, int64_t pts) 3555{ 3556 audio_track_cblk_t* cblk = this->cblk(); 3557 uint32_t framesReady; 3558 uint32_t framesReq = buffer->frameCount; 3559 3560 // Check if last stepServer failed, try to step now 3561 if (mStepServerFailed) { 3562 if (!step()) goto getNextBuffer_exit; 3563 ALOGV("stepServer recovered"); 3564 mStepServerFailed = false; 3565 } 3566 3567 framesReady = cblk->framesReady(); 3568 3569 if (CC_LIKELY(framesReady)) { 3570 uint32_t s = cblk->server; 3571 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3572 3573 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3574 if (framesReq > framesReady) { 3575 framesReq = framesReady; 3576 } 3577 if (s + framesReq > bufferEnd) { 3578 framesReq = bufferEnd - s; 3579 } 3580 3581 buffer->raw = getBuffer(s, framesReq); 3582 if (buffer->raw == NULL) goto getNextBuffer_exit; 3583 3584 buffer->frameCount = framesReq; 3585 return NO_ERROR; 3586 } 3587 3588getNextBuffer_exit: 3589 buffer->raw = NULL; 3590 buffer->frameCount = 0; 3591 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3592 return NOT_ENOUGH_DATA; 3593} 3594 3595uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3596 return mCblk->framesReady(); 3597} 3598 3599bool AudioFlinger::PlaybackThread::Track::isReady() const { 3600 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3601 3602 if (framesReady() >= mCblk->frameCount || 3603 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3604 mFillingUpStatus = FS_FILLED; 3605 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3606 return true; 3607 } 3608 return false; 3609} 3610 3611status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3612{ 3613 status_t status = NO_ERROR; 3614 ALOGV("start(%d), calling pid %d session %d tid %d", 3615 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3616 sp<ThreadBase> thread = mThread.promote(); 3617 if (thread != 0) { 3618 Mutex::Autolock _l(thread->mLock); 3619 track_state state = mState; 3620 // here the track could be either new, or restarted 3621 // in both cases "unstop" the track 3622 if (mState == PAUSED) { 3623 mState = TrackBase::RESUMING; 3624 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3625 } else { 3626 mState = TrackBase::ACTIVE; 3627 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3628 } 3629 3630 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3631 thread->mLock.unlock(); 3632 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3633 thread->mLock.lock(); 3634 3635 // to track the speaker usage 3636 if (status == NO_ERROR) { 3637 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3638 } 3639 } 3640 if (status == NO_ERROR) { 3641 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3642 playbackThread->addTrack_l(this); 3643 } else { 3644 mState = state; 3645 } 3646 } else { 3647 status = BAD_VALUE; 3648 } 3649 return status; 3650} 3651 3652void AudioFlinger::PlaybackThread::Track::stop() 3653{ 3654 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3655 sp<ThreadBase> thread = mThread.promote(); 3656 if (thread != 0) { 3657 Mutex::Autolock _l(thread->mLock); 3658 track_state state = mState; 3659 if (mState > STOPPED) { 3660 mState = STOPPED; 3661 // If the track is not active (PAUSED and buffers full), flush buffers 3662 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3663 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3664 reset(); 3665 } 3666 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3667 } 3668 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3669 thread->mLock.unlock(); 3670 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3671 thread->mLock.lock(); 3672 3673 // to track the speaker usage 3674 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3675 } 3676 } 3677} 3678 3679void AudioFlinger::PlaybackThread::Track::pause() 3680{ 3681 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3682 sp<ThreadBase> thread = mThread.promote(); 3683 if (thread != 0) { 3684 Mutex::Autolock _l(thread->mLock); 3685 if (mState == ACTIVE || mState == RESUMING) { 3686 mState = PAUSING; 3687 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3688 if (!isOutputTrack()) { 3689 thread->mLock.unlock(); 3690 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3691 thread->mLock.lock(); 3692 3693 // to track the speaker usage 3694 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3695 } 3696 } 3697 } 3698} 3699 3700void AudioFlinger::PlaybackThread::Track::flush() 3701{ 3702 ALOGV("flush(%d)", mName); 3703 sp<ThreadBase> thread = mThread.promote(); 3704 if (thread != 0) { 3705 Mutex::Autolock _l(thread->mLock); 3706 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3707 return; 3708 } 3709 // No point remaining in PAUSED state after a flush => go to 3710 // STOPPED state 3711 mState = STOPPED; 3712 3713 // do not reset the track if it is still in the process of being stopped or paused. 3714 // this will be done by prepareTracks_l() when the track is stopped. 3715 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3716 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3717 reset(); 3718 } 3719 } 3720} 3721 3722void AudioFlinger::PlaybackThread::Track::reset() 3723{ 3724 // Do not reset twice to avoid discarding data written just after a flush and before 3725 // the audioflinger thread detects the track is stopped. 3726 if (!mResetDone) { 3727 TrackBase::reset(); 3728 // Force underrun condition to avoid false underrun callback until first data is 3729 // written to buffer 3730 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3731 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3732 mFillingUpStatus = FS_FILLING; 3733 mResetDone = true; 3734 } 3735} 3736 3737void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3738{ 3739 mMute = muted; 3740} 3741 3742status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3743{ 3744 status_t status = DEAD_OBJECT; 3745 sp<ThreadBase> thread = mThread.promote(); 3746 if (thread != 0) { 3747 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3748 status = playbackThread->attachAuxEffect(this, EffectId); 3749 } 3750 return status; 3751} 3752 3753void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3754{ 3755 mAuxEffectId = EffectId; 3756 mAuxBuffer = buffer; 3757} 3758 3759// timed audio tracks 3760 3761sp<AudioFlinger::PlaybackThread::TimedTrack> 3762AudioFlinger::PlaybackThread::TimedTrack::create( 3763 PlaybackThread *thread, 3764 const sp<Client>& client, 3765 audio_stream_type_t streamType, 3766 uint32_t sampleRate, 3767 audio_format_t format, 3768 uint32_t channelMask, 3769 int frameCount, 3770 const sp<IMemory>& sharedBuffer, 3771 int sessionId) { 3772 if (!client->reserveTimedTrack()) 3773 return NULL; 3774 3775 sp<TimedTrack> track = new TimedTrack( 3776 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3777 sharedBuffer, sessionId); 3778 3779 if (track == NULL) { 3780 client->releaseTimedTrack(); 3781 return NULL; 3782 } 3783 3784 return track; 3785} 3786 3787AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3788 PlaybackThread *thread, 3789 const sp<Client>& client, 3790 audio_stream_type_t streamType, 3791 uint32_t sampleRate, 3792 audio_format_t format, 3793 uint32_t channelMask, 3794 int frameCount, 3795 const sp<IMemory>& sharedBuffer, 3796 int sessionId) 3797 : Track(thread, client, streamType, sampleRate, format, channelMask, 3798 frameCount, sharedBuffer, sessionId), 3799 mTimedSilenceBuffer(NULL), 3800 mTimedSilenceBufferSize(0), 3801 mTimedAudioOutputOnTime(false), 3802 mMediaTimeTransformValid(false) 3803{ 3804 LocalClock lc; 3805 mLocalTimeFreq = lc.getLocalFreq(); 3806 3807 mLocalTimeToSampleTransform.a_zero = 0; 3808 mLocalTimeToSampleTransform.b_zero = 0; 3809 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3810 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3811 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3812 &mLocalTimeToSampleTransform.a_to_b_denom); 3813} 3814 3815AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3816 mClient->releaseTimedTrack(); 3817 delete [] mTimedSilenceBuffer; 3818} 3819 3820status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3821 size_t size, sp<IMemory>* buffer) { 3822 3823 Mutex::Autolock _l(mTimedBufferQueueLock); 3824 3825 trimTimedBufferQueue_l(); 3826 3827 // lazily initialize the shared memory heap for timed buffers 3828 if (mTimedMemoryDealer == NULL) { 3829 const int kTimedBufferHeapSize = 512 << 10; 3830 3831 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3832 "AudioFlingerTimed"); 3833 if (mTimedMemoryDealer == NULL) 3834 return NO_MEMORY; 3835 } 3836 3837 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3838 if (newBuffer == NULL) { 3839 newBuffer = mTimedMemoryDealer->allocate(size); 3840 if (newBuffer == NULL) 3841 return NO_MEMORY; 3842 } 3843 3844 *buffer = newBuffer; 3845 return NO_ERROR; 3846} 3847 3848// caller must hold mTimedBufferQueueLock 3849void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3850 int64_t mediaTimeNow; 3851 { 3852 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3853 if (!mMediaTimeTransformValid) 3854 return; 3855 3856 int64_t targetTimeNow; 3857 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3858 ? mCCHelper.getCommonTime(&targetTimeNow) 3859 : mCCHelper.getLocalTime(&targetTimeNow); 3860 3861 if (OK != res) 3862 return; 3863 3864 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3865 &mediaTimeNow)) { 3866 return; 3867 } 3868 } 3869 3870 size_t trimIndex; 3871 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3872 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3873 break; 3874 } 3875 3876 if (trimIndex) { 3877 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3878 } 3879} 3880 3881status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3882 const sp<IMemory>& buffer, int64_t pts) { 3883 3884 { 3885 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3886 if (!mMediaTimeTransformValid) 3887 return INVALID_OPERATION; 3888 } 3889 3890 Mutex::Autolock _l(mTimedBufferQueueLock); 3891 3892 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3893 3894 return NO_ERROR; 3895} 3896 3897status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3898 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3899 3900 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3901 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3902 target); 3903 3904 if (!(target == TimedAudioTrack::LOCAL_TIME || 3905 target == TimedAudioTrack::COMMON_TIME)) { 3906 return BAD_VALUE; 3907 } 3908 3909 Mutex::Autolock lock(mMediaTimeTransformLock); 3910 mMediaTimeTransform = xform; 3911 mMediaTimeTransformTarget = target; 3912 mMediaTimeTransformValid = true; 3913 3914 return NO_ERROR; 3915} 3916 3917#define min(a, b) ((a) < (b) ? (a) : (b)) 3918 3919// implementation of getNextBuffer for tracks whose buffers have timestamps 3920status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3921 AudioBufferProvider::Buffer* buffer, int64_t pts) 3922{ 3923 if (pts == AudioBufferProvider::kInvalidPTS) { 3924 buffer->raw = 0; 3925 buffer->frameCount = 0; 3926 return INVALID_OPERATION; 3927 } 3928 3929 Mutex::Autolock _l(mTimedBufferQueueLock); 3930 3931 while (true) { 3932 3933 // if we have no timed buffers, then fail 3934 if (mTimedBufferQueue.isEmpty()) { 3935 buffer->raw = 0; 3936 buffer->frameCount = 0; 3937 return NOT_ENOUGH_DATA; 3938 } 3939 3940 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3941 3942 // calculate the PTS of the head of the timed buffer queue expressed in 3943 // local time 3944 int64_t headLocalPTS; 3945 { 3946 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3947 3948 assert(mMediaTimeTransformValid); 3949 3950 if (mMediaTimeTransform.a_to_b_denom == 0) { 3951 // the transform represents a pause, so yield silence 3952 timedYieldSilence(buffer->frameCount, buffer); 3953 return NO_ERROR; 3954 } 3955 3956 int64_t transformedPTS; 3957 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3958 &transformedPTS)) { 3959 // the transform failed. this shouldn't happen, but if it does 3960 // then just drop this buffer 3961 ALOGW("timedGetNextBuffer transform failed"); 3962 buffer->raw = 0; 3963 buffer->frameCount = 0; 3964 mTimedBufferQueue.removeAt(0); 3965 return NO_ERROR; 3966 } 3967 3968 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3969 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3970 &headLocalPTS)) { 3971 buffer->raw = 0; 3972 buffer->frameCount = 0; 3973 return INVALID_OPERATION; 3974 } 3975 } else { 3976 headLocalPTS = transformedPTS; 3977 } 3978 } 3979 3980 // adjust the head buffer's PTS to reflect the portion of the head buffer 3981 // that has already been consumed 3982 int64_t effectivePTS = headLocalPTS + 3983 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3984 3985 // Calculate the delta in samples between the head of the input buffer 3986 // queue and the start of the next output buffer that will be written. 3987 // If the transformation fails because of over or underflow, it means 3988 // that the sample's position in the output stream is so far out of 3989 // whack that it should just be dropped. 3990 int64_t sampleDelta; 3991 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3992 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3993 mTimedBufferQueue.removeAt(0); 3994 continue; 3995 } 3996 if (!mLocalTimeToSampleTransform.doForwardTransform( 3997 (effectivePTS - pts) << 32, &sampleDelta)) { 3998 ALOGV("*** too late during sample rate transform: dropped buffer"); 3999 mTimedBufferQueue.removeAt(0); 4000 continue; 4001 } 4002 4003 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4004 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4005 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4006 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4007 4008 // if the delta between the ideal placement for the next input sample and 4009 // the current output position is within this threshold, then we will 4010 // concatenate the next input samples to the previous output 4011 const int64_t kSampleContinuityThreshold = 4012 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4013 4014 // if this is the first buffer of audio that we're emitting from this track 4015 // then it should be almost exactly on time. 4016 const int64_t kSampleStartupThreshold = 1LL << 32; 4017 4018 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4019 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4020 // the next input is close enough to being on time, so concatenate it 4021 // with the last output 4022 timedYieldSamples(buffer); 4023 4024 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4025 return NO_ERROR; 4026 } else if (sampleDelta > 0) { 4027 // the gap between the current output position and the proper start of 4028 // the next input sample is too big, so fill it with silence 4029 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4030 4031 timedYieldSilence(framesUntilNextInput, buffer); 4032 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4033 return NO_ERROR; 4034 } else { 4035 // the next input sample is late 4036 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4037 size_t onTimeSamplePosition = 4038 head.position() + lateFrames * mCblk->frameSize; 4039 4040 if (onTimeSamplePosition > head.buffer()->size()) { 4041 // all the remaining samples in the head are too late, so 4042 // drop it and move on 4043 ALOGV("*** too late: dropped buffer"); 4044 mTimedBufferQueue.removeAt(0); 4045 continue; 4046 } else { 4047 // skip over the late samples 4048 head.setPosition(onTimeSamplePosition); 4049 4050 // yield the available samples 4051 timedYieldSamples(buffer); 4052 4053 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4054 return NO_ERROR; 4055 } 4056 } 4057 } 4058} 4059 4060// Yield samples from the timed buffer queue head up to the given output 4061// buffer's capacity. 4062// 4063// Caller must hold mTimedBufferQueueLock 4064void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4065 AudioBufferProvider::Buffer* buffer) { 4066 4067 const TimedBuffer& head = mTimedBufferQueue[0]; 4068 4069 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4070 head.position()); 4071 4072 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4073 mCblk->frameSize); 4074 size_t framesRequested = buffer->frameCount; 4075 buffer->frameCount = min(framesLeftInHead, framesRequested); 4076 4077 mTimedAudioOutputOnTime = true; 4078} 4079 4080// Yield samples of silence up to the given output buffer's capacity 4081// 4082// Caller must hold mTimedBufferQueueLock 4083void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4084 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4085 4086 // lazily allocate a buffer filled with silence 4087 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4088 delete [] mTimedSilenceBuffer; 4089 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4090 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4091 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4092 } 4093 4094 buffer->raw = mTimedSilenceBuffer; 4095 size_t framesRequested = buffer->frameCount; 4096 buffer->frameCount = min(numFrames, framesRequested); 4097 4098 mTimedAudioOutputOnTime = false; 4099} 4100 4101void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4102 AudioBufferProvider::Buffer* buffer) { 4103 4104 Mutex::Autolock _l(mTimedBufferQueueLock); 4105 4106 // If the buffer which was just released is part of the buffer at the head 4107 // of the queue, be sure to update the amt of the buffer which has been 4108 // consumed. If the buffer being returned is not part of the head of the 4109 // queue, its either because the buffer is part of the silence buffer, or 4110 // because the head of the timed queue was trimmed after the mixer called 4111 // getNextBuffer but before the mixer called releaseBuffer. 4112 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4113 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4114 4115 void* start = head.buffer()->pointer(); 4116 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4117 4118 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4119 head.setPosition(head.position() + 4120 (buffer->frameCount * mCblk->frameSize)); 4121 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4122 mTimedBufferQueue.removeAt(0); 4123 } 4124 } 4125 } 4126 4127 buffer->raw = 0; 4128 buffer->frameCount = 0; 4129} 4130 4131uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4132 Mutex::Autolock _l(mTimedBufferQueueLock); 4133 4134 uint32_t frames = 0; 4135 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4136 const TimedBuffer& tb = mTimedBufferQueue[i]; 4137 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4138 } 4139 4140 return frames; 4141} 4142 4143AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4144 : mPTS(0), mPosition(0) {} 4145 4146AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4147 const sp<IMemory>& buffer, int64_t pts) 4148 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4149 4150// ---------------------------------------------------------------------------- 4151 4152// RecordTrack constructor must be called with AudioFlinger::mLock held 4153AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4154 RecordThread *thread, 4155 const sp<Client>& client, 4156 uint32_t sampleRate, 4157 audio_format_t format, 4158 uint32_t channelMask, 4159 int frameCount, 4160 int sessionId) 4161 : TrackBase(thread, client, sampleRate, format, 4162 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4163 mOverflow(false) 4164{ 4165 if (mCblk != NULL) { 4166 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4167 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4168 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4169 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4170 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4171 } else { 4172 mCblk->frameSize = sizeof(int8_t); 4173 } 4174 } 4175} 4176 4177AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4178{ 4179 sp<ThreadBase> thread = mThread.promote(); 4180 if (thread != 0) { 4181 AudioSystem::releaseInput(thread->id()); 4182 } 4183} 4184 4185status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4186{ 4187 audio_track_cblk_t* cblk = this->cblk(); 4188 uint32_t framesAvail; 4189 uint32_t framesReq = buffer->frameCount; 4190 4191 // Check if last stepServer failed, try to step now 4192 if (mStepServerFailed) { 4193 if (!step()) goto getNextBuffer_exit; 4194 ALOGV("stepServer recovered"); 4195 mStepServerFailed = false; 4196 } 4197 4198 framesAvail = cblk->framesAvailable_l(); 4199 4200 if (CC_LIKELY(framesAvail)) { 4201 uint32_t s = cblk->server; 4202 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4203 4204 if (framesReq > framesAvail) { 4205 framesReq = framesAvail; 4206 } 4207 if (s + framesReq > bufferEnd) { 4208 framesReq = bufferEnd - s; 4209 } 4210 4211 buffer->raw = getBuffer(s, framesReq); 4212 if (buffer->raw == NULL) goto getNextBuffer_exit; 4213 4214 buffer->frameCount = framesReq; 4215 return NO_ERROR; 4216 } 4217 4218getNextBuffer_exit: 4219 buffer->raw = NULL; 4220 buffer->frameCount = 0; 4221 return NOT_ENOUGH_DATA; 4222} 4223 4224status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4225{ 4226 sp<ThreadBase> thread = mThread.promote(); 4227 if (thread != 0) { 4228 RecordThread *recordThread = (RecordThread *)thread.get(); 4229 return recordThread->start(this, tid); 4230 } else { 4231 return BAD_VALUE; 4232 } 4233} 4234 4235void AudioFlinger::RecordThread::RecordTrack::stop() 4236{ 4237 sp<ThreadBase> thread = mThread.promote(); 4238 if (thread != 0) { 4239 RecordThread *recordThread = (RecordThread *)thread.get(); 4240 recordThread->stop(this); 4241 TrackBase::reset(); 4242 // Force overerrun condition to avoid false overrun callback until first data is 4243 // read from buffer 4244 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4245 } 4246} 4247 4248void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4249{ 4250 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4251 (mClient == 0) ? getpid_cached : mClient->pid(), 4252 mFormat, 4253 mChannelMask, 4254 mSessionId, 4255 mFrameCount, 4256 mState, 4257 mCblk->sampleRate, 4258 mCblk->server, 4259 mCblk->user); 4260} 4261 4262 4263// ---------------------------------------------------------------------------- 4264 4265AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4266 PlaybackThread *playbackThread, 4267 DuplicatingThread *sourceThread, 4268 uint32_t sampleRate, 4269 audio_format_t format, 4270 uint32_t channelMask, 4271 int frameCount) 4272 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4273 mActive(false), mSourceThread(sourceThread) 4274{ 4275 4276 if (mCblk != NULL) { 4277 mCblk->flags |= CBLK_DIRECTION_OUT; 4278 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4279 mOutBuffer.frameCount = 0; 4280 playbackThread->mTracks.add(this); 4281 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4282 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4283 mCblk, mBuffer, mCblk->buffers, 4284 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4285 } else { 4286 ALOGW("Error creating output track on thread %p", playbackThread); 4287 } 4288} 4289 4290AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4291{ 4292 clearBufferQueue(); 4293} 4294 4295status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4296{ 4297 status_t status = Track::start(tid); 4298 if (status != NO_ERROR) { 4299 return status; 4300 } 4301 4302 mActive = true; 4303 mRetryCount = 127; 4304 return status; 4305} 4306 4307void AudioFlinger::PlaybackThread::OutputTrack::stop() 4308{ 4309 Track::stop(); 4310 clearBufferQueue(); 4311 mOutBuffer.frameCount = 0; 4312 mActive = false; 4313} 4314 4315bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4316{ 4317 Buffer *pInBuffer; 4318 Buffer inBuffer; 4319 uint32_t channelCount = mChannelCount; 4320 bool outputBufferFull = false; 4321 inBuffer.frameCount = frames; 4322 inBuffer.i16 = data; 4323 4324 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4325 4326 if (!mActive && frames != 0) { 4327 start(0); 4328 sp<ThreadBase> thread = mThread.promote(); 4329 if (thread != 0) { 4330 MixerThread *mixerThread = (MixerThread *)thread.get(); 4331 if (mCblk->frameCount > frames){ 4332 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4333 uint32_t startFrames = (mCblk->frameCount - frames); 4334 pInBuffer = new Buffer; 4335 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4336 pInBuffer->frameCount = startFrames; 4337 pInBuffer->i16 = pInBuffer->mBuffer; 4338 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4339 mBufferQueue.add(pInBuffer); 4340 } else { 4341 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4342 } 4343 } 4344 } 4345 } 4346 4347 while (waitTimeLeftMs) { 4348 // First write pending buffers, then new data 4349 if (mBufferQueue.size()) { 4350 pInBuffer = mBufferQueue.itemAt(0); 4351 } else { 4352 pInBuffer = &inBuffer; 4353 } 4354 4355 if (pInBuffer->frameCount == 0) { 4356 break; 4357 } 4358 4359 if (mOutBuffer.frameCount == 0) { 4360 mOutBuffer.frameCount = pInBuffer->frameCount; 4361 nsecs_t startTime = systemTime(); 4362 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4363 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4364 outputBufferFull = true; 4365 break; 4366 } 4367 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4368 if (waitTimeLeftMs >= waitTimeMs) { 4369 waitTimeLeftMs -= waitTimeMs; 4370 } else { 4371 waitTimeLeftMs = 0; 4372 } 4373 } 4374 4375 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4376 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4377 mCblk->stepUser(outFrames); 4378 pInBuffer->frameCount -= outFrames; 4379 pInBuffer->i16 += outFrames * channelCount; 4380 mOutBuffer.frameCount -= outFrames; 4381 mOutBuffer.i16 += outFrames * channelCount; 4382 4383 if (pInBuffer->frameCount == 0) { 4384 if (mBufferQueue.size()) { 4385 mBufferQueue.removeAt(0); 4386 delete [] pInBuffer->mBuffer; 4387 delete pInBuffer; 4388 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4389 } else { 4390 break; 4391 } 4392 } 4393 } 4394 4395 // If we could not write all frames, allocate a buffer and queue it for next time. 4396 if (inBuffer.frameCount) { 4397 sp<ThreadBase> thread = mThread.promote(); 4398 if (thread != 0 && !thread->standby()) { 4399 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4400 pInBuffer = new Buffer; 4401 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4402 pInBuffer->frameCount = inBuffer.frameCount; 4403 pInBuffer->i16 = pInBuffer->mBuffer; 4404 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4405 mBufferQueue.add(pInBuffer); 4406 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4407 } else { 4408 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4409 } 4410 } 4411 } 4412 4413 // Calling write() with a 0 length buffer, means that no more data will be written: 4414 // If no more buffers are pending, fill output track buffer to make sure it is started 4415 // by output mixer. 4416 if (frames == 0 && mBufferQueue.size() == 0) { 4417 if (mCblk->user < mCblk->frameCount) { 4418 frames = mCblk->frameCount - mCblk->user; 4419 pInBuffer = new Buffer; 4420 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4421 pInBuffer->frameCount = frames; 4422 pInBuffer->i16 = pInBuffer->mBuffer; 4423 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4424 mBufferQueue.add(pInBuffer); 4425 } else if (mActive) { 4426 stop(); 4427 } 4428 } 4429 4430 return outputBufferFull; 4431} 4432 4433status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4434{ 4435 int active; 4436 status_t result; 4437 audio_track_cblk_t* cblk = mCblk; 4438 uint32_t framesReq = buffer->frameCount; 4439 4440// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4441 buffer->frameCount = 0; 4442 4443 uint32_t framesAvail = cblk->framesAvailable(); 4444 4445 4446 if (framesAvail == 0) { 4447 Mutex::Autolock _l(cblk->lock); 4448 goto start_loop_here; 4449 while (framesAvail == 0) { 4450 active = mActive; 4451 if (CC_UNLIKELY(!active)) { 4452 ALOGV("Not active and NO_MORE_BUFFERS"); 4453 return NO_MORE_BUFFERS; 4454 } 4455 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4456 if (result != NO_ERROR) { 4457 return NO_MORE_BUFFERS; 4458 } 4459 // read the server count again 4460 start_loop_here: 4461 framesAvail = cblk->framesAvailable_l(); 4462 } 4463 } 4464 4465// if (framesAvail < framesReq) { 4466// return NO_MORE_BUFFERS; 4467// } 4468 4469 if (framesReq > framesAvail) { 4470 framesReq = framesAvail; 4471 } 4472 4473 uint32_t u = cblk->user; 4474 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4475 4476 if (u + framesReq > bufferEnd) { 4477 framesReq = bufferEnd - u; 4478 } 4479 4480 buffer->frameCount = framesReq; 4481 buffer->raw = (void *)cblk->buffer(u); 4482 return NO_ERROR; 4483} 4484 4485 4486void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4487{ 4488 size_t size = mBufferQueue.size(); 4489 4490 for (size_t i = 0; i < size; i++) { 4491 Buffer *pBuffer = mBufferQueue.itemAt(i); 4492 delete [] pBuffer->mBuffer; 4493 delete pBuffer; 4494 } 4495 mBufferQueue.clear(); 4496} 4497 4498// ---------------------------------------------------------------------------- 4499 4500AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4501 : RefBase(), 4502 mAudioFlinger(audioFlinger), 4503 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4504 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4505 mPid(pid), 4506 mTimedTrackCount(0) 4507{ 4508 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4509} 4510 4511// Client destructor must be called with AudioFlinger::mLock held 4512AudioFlinger::Client::~Client() 4513{ 4514 mAudioFlinger->removeClient_l(mPid); 4515} 4516 4517sp<MemoryDealer> AudioFlinger::Client::heap() const 4518{ 4519 return mMemoryDealer; 4520} 4521 4522// Reserve one of the limited slots for a timed audio track associated 4523// with this client 4524bool AudioFlinger::Client::reserveTimedTrack() 4525{ 4526 const int kMaxTimedTracksPerClient = 4; 4527 4528 Mutex::Autolock _l(mTimedTrackLock); 4529 4530 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4531 ALOGW("can not create timed track - pid %d has exceeded the limit", 4532 mPid); 4533 return false; 4534 } 4535 4536 mTimedTrackCount++; 4537 return true; 4538} 4539 4540// Release a slot for a timed audio track 4541void AudioFlinger::Client::releaseTimedTrack() 4542{ 4543 Mutex::Autolock _l(mTimedTrackLock); 4544 mTimedTrackCount--; 4545} 4546 4547// ---------------------------------------------------------------------------- 4548 4549AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4550 const sp<IAudioFlingerClient>& client, 4551 pid_t pid) 4552 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4553{ 4554} 4555 4556AudioFlinger::NotificationClient::~NotificationClient() 4557{ 4558} 4559 4560void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4561{ 4562 sp<NotificationClient> keep(this); 4563 mAudioFlinger->removeNotificationClient(mPid); 4564} 4565 4566// ---------------------------------------------------------------------------- 4567 4568AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4569 : BnAudioTrack(), 4570 mTrack(track) 4571{ 4572} 4573 4574AudioFlinger::TrackHandle::~TrackHandle() { 4575 // just stop the track on deletion, associated resources 4576 // will be freed from the main thread once all pending buffers have 4577 // been played. Unless it's not in the active track list, in which 4578 // case we free everything now... 4579 mTrack->destroy(); 4580} 4581 4582sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4583 return mTrack->getCblk(); 4584} 4585 4586status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4587 return mTrack->start(tid); 4588} 4589 4590void AudioFlinger::TrackHandle::stop() { 4591 mTrack->stop(); 4592} 4593 4594void AudioFlinger::TrackHandle::flush() { 4595 mTrack->flush(); 4596} 4597 4598void AudioFlinger::TrackHandle::mute(bool e) { 4599 mTrack->mute(e); 4600} 4601 4602void AudioFlinger::TrackHandle::pause() { 4603 mTrack->pause(); 4604} 4605 4606status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4607{ 4608 return mTrack->attachAuxEffect(EffectId); 4609} 4610 4611status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4612 sp<IMemory>* buffer) { 4613 if (!mTrack->isTimedTrack()) 4614 return INVALID_OPERATION; 4615 4616 PlaybackThread::TimedTrack* tt = 4617 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4618 return tt->allocateTimedBuffer(size, buffer); 4619} 4620 4621status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4622 int64_t pts) { 4623 if (!mTrack->isTimedTrack()) 4624 return INVALID_OPERATION; 4625 4626 PlaybackThread::TimedTrack* tt = 4627 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4628 return tt->queueTimedBuffer(buffer, pts); 4629} 4630 4631status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4632 const LinearTransform& xform, int target) { 4633 4634 if (!mTrack->isTimedTrack()) 4635 return INVALID_OPERATION; 4636 4637 PlaybackThread::TimedTrack* tt = 4638 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4639 return tt->setMediaTimeTransform( 4640 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4641} 4642 4643status_t AudioFlinger::TrackHandle::onTransact( 4644 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4645{ 4646 return BnAudioTrack::onTransact(code, data, reply, flags); 4647} 4648 4649// ---------------------------------------------------------------------------- 4650 4651sp<IAudioRecord> AudioFlinger::openRecord( 4652 pid_t pid, 4653 audio_io_handle_t input, 4654 uint32_t sampleRate, 4655 audio_format_t format, 4656 uint32_t channelMask, 4657 int frameCount, 4658 // FIXME dead, remove from IAudioFlinger 4659 uint32_t flags, 4660 int *sessionId, 4661 status_t *status) 4662{ 4663 sp<RecordThread::RecordTrack> recordTrack; 4664 sp<RecordHandle> recordHandle; 4665 sp<Client> client; 4666 status_t lStatus; 4667 RecordThread *thread; 4668 size_t inFrameCount; 4669 int lSessionId; 4670 4671 // check calling permissions 4672 if (!recordingAllowed()) { 4673 lStatus = PERMISSION_DENIED; 4674 goto Exit; 4675 } 4676 4677 // add client to list 4678 { // scope for mLock 4679 Mutex::Autolock _l(mLock); 4680 thread = checkRecordThread_l(input); 4681 if (thread == NULL) { 4682 lStatus = BAD_VALUE; 4683 goto Exit; 4684 } 4685 4686 client = registerPid_l(pid); 4687 4688 // If no audio session id is provided, create one here 4689 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4690 lSessionId = *sessionId; 4691 } else { 4692 lSessionId = nextUniqueId(); 4693 if (sessionId != NULL) { 4694 *sessionId = lSessionId; 4695 } 4696 } 4697 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4698 recordTrack = thread->createRecordTrack_l(client, 4699 sampleRate, 4700 format, 4701 channelMask, 4702 frameCount, 4703 lSessionId, 4704 &lStatus); 4705 } 4706 if (lStatus != NO_ERROR) { 4707 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4708 // destructor is called by the TrackBase destructor with mLock held 4709 client.clear(); 4710 recordTrack.clear(); 4711 goto Exit; 4712 } 4713 4714 // return to handle to client 4715 recordHandle = new RecordHandle(recordTrack); 4716 lStatus = NO_ERROR; 4717 4718Exit: 4719 if (status) { 4720 *status = lStatus; 4721 } 4722 return recordHandle; 4723} 4724 4725// ---------------------------------------------------------------------------- 4726 4727AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4728 : BnAudioRecord(), 4729 mRecordTrack(recordTrack) 4730{ 4731} 4732 4733AudioFlinger::RecordHandle::~RecordHandle() { 4734 stop(); 4735} 4736 4737sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4738 return mRecordTrack->getCblk(); 4739} 4740 4741status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4742 ALOGV("RecordHandle::start()"); 4743 return mRecordTrack->start(tid); 4744} 4745 4746void AudioFlinger::RecordHandle::stop() { 4747 ALOGV("RecordHandle::stop()"); 4748 mRecordTrack->stop(); 4749} 4750 4751status_t AudioFlinger::RecordHandle::onTransact( 4752 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4753{ 4754 return BnAudioRecord::onTransact(code, data, reply, flags); 4755} 4756 4757// ---------------------------------------------------------------------------- 4758 4759AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4760 AudioStreamIn *input, 4761 uint32_t sampleRate, 4762 uint32_t channels, 4763 audio_io_handle_t id, 4764 uint32_t device) : 4765 ThreadBase(audioFlinger, id, device, RECORD), 4766 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4767 // mRsmpInIndex and mInputBytes set by readInputParameters() 4768 mReqChannelCount(popcount(channels)), 4769 mReqSampleRate(sampleRate) 4770 // mBytesRead is only meaningful while active, and so is cleared in start() 4771 // (but might be better to also clear here for dump?) 4772{ 4773 snprintf(mName, kNameLength, "AudioIn_%d", id); 4774 4775 readInputParameters(); 4776} 4777 4778 4779AudioFlinger::RecordThread::~RecordThread() 4780{ 4781 delete[] mRsmpInBuffer; 4782 delete mResampler; 4783 delete[] mRsmpOutBuffer; 4784} 4785 4786void AudioFlinger::RecordThread::onFirstRef() 4787{ 4788 run(mName, PRIORITY_URGENT_AUDIO); 4789} 4790 4791status_t AudioFlinger::RecordThread::readyToRun() 4792{ 4793 status_t status = initCheck(); 4794 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4795 return status; 4796} 4797 4798bool AudioFlinger::RecordThread::threadLoop() 4799{ 4800 AudioBufferProvider::Buffer buffer; 4801 sp<RecordTrack> activeTrack; 4802 Vector< sp<EffectChain> > effectChains; 4803 4804 nsecs_t lastWarning = 0; 4805 4806 acquireWakeLock(); 4807 4808 // start recording 4809 while (!exitPending()) { 4810 4811 processConfigEvents(); 4812 4813 { // scope for mLock 4814 Mutex::Autolock _l(mLock); 4815 checkForNewParameters_l(); 4816 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4817 if (!mStandby) { 4818 mInput->stream->common.standby(&mInput->stream->common); 4819 mStandby = true; 4820 } 4821 4822 if (exitPending()) break; 4823 4824 releaseWakeLock_l(); 4825 ALOGV("RecordThread: loop stopping"); 4826 // go to sleep 4827 mWaitWorkCV.wait(mLock); 4828 ALOGV("RecordThread: loop starting"); 4829 acquireWakeLock_l(); 4830 continue; 4831 } 4832 if (mActiveTrack != 0) { 4833 if (mActiveTrack->mState == TrackBase::PAUSING) { 4834 if (!mStandby) { 4835 mInput->stream->common.standby(&mInput->stream->common); 4836 mStandby = true; 4837 } 4838 mActiveTrack.clear(); 4839 mStartStopCond.broadcast(); 4840 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4841 if (mReqChannelCount != mActiveTrack->channelCount()) { 4842 mActiveTrack.clear(); 4843 mStartStopCond.broadcast(); 4844 } else if (mBytesRead != 0) { 4845 // record start succeeds only if first read from audio input 4846 // succeeds 4847 if (mBytesRead > 0) { 4848 mActiveTrack->mState = TrackBase::ACTIVE; 4849 } else { 4850 mActiveTrack.clear(); 4851 } 4852 mStartStopCond.broadcast(); 4853 } 4854 mStandby = false; 4855 } 4856 } 4857 lockEffectChains_l(effectChains); 4858 } 4859 4860 if (mActiveTrack != 0) { 4861 if (mActiveTrack->mState != TrackBase::ACTIVE && 4862 mActiveTrack->mState != TrackBase::RESUMING) { 4863 unlockEffectChains(effectChains); 4864 usleep(kRecordThreadSleepUs); 4865 continue; 4866 } 4867 for (size_t i = 0; i < effectChains.size(); i ++) { 4868 effectChains[i]->process_l(); 4869 } 4870 4871 buffer.frameCount = mFrameCount; 4872 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4873 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4874 size_t framesOut = buffer.frameCount; 4875 if (mResampler == NULL) { 4876 // no resampling 4877 while (framesOut) { 4878 size_t framesIn = mFrameCount - mRsmpInIndex; 4879 if (framesIn) { 4880 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4881 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4882 if (framesIn > framesOut) 4883 framesIn = framesOut; 4884 mRsmpInIndex += framesIn; 4885 framesOut -= framesIn; 4886 if ((int)mChannelCount == mReqChannelCount || 4887 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4888 memcpy(dst, src, framesIn * mFrameSize); 4889 } else { 4890 int16_t *src16 = (int16_t *)src; 4891 int16_t *dst16 = (int16_t *)dst; 4892 if (mChannelCount == 1) { 4893 while (framesIn--) { 4894 *dst16++ = *src16; 4895 *dst16++ = *src16++; 4896 } 4897 } else { 4898 while (framesIn--) { 4899 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4900 src16 += 2; 4901 } 4902 } 4903 } 4904 } 4905 if (framesOut && mFrameCount == mRsmpInIndex) { 4906 if (framesOut == mFrameCount && 4907 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4908 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4909 framesOut = 0; 4910 } else { 4911 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4912 mRsmpInIndex = 0; 4913 } 4914 if (mBytesRead < 0) { 4915 ALOGE("Error reading audio input"); 4916 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4917 // Force input into standby so that it tries to 4918 // recover at next read attempt 4919 mInput->stream->common.standby(&mInput->stream->common); 4920 usleep(kRecordThreadSleepUs); 4921 } 4922 mRsmpInIndex = mFrameCount; 4923 framesOut = 0; 4924 buffer.frameCount = 0; 4925 } 4926 } 4927 } 4928 } else { 4929 // resampling 4930 4931 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4932 // alter output frame count as if we were expecting stereo samples 4933 if (mChannelCount == 1 && mReqChannelCount == 1) { 4934 framesOut >>= 1; 4935 } 4936 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4937 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4938 // are 32 bit aligned which should be always true. 4939 if (mChannelCount == 2 && mReqChannelCount == 1) { 4940 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4941 // the resampler always outputs stereo samples: do post stereo to mono conversion 4942 int16_t *src = (int16_t *)mRsmpOutBuffer; 4943 int16_t *dst = buffer.i16; 4944 while (framesOut--) { 4945 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4946 src += 2; 4947 } 4948 } else { 4949 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4950 } 4951 4952 } 4953 mActiveTrack->releaseBuffer(&buffer); 4954 mActiveTrack->overflow(); 4955 } 4956 // client isn't retrieving buffers fast enough 4957 else { 4958 if (!mActiveTrack->setOverflow()) { 4959 nsecs_t now = systemTime(); 4960 if ((now - lastWarning) > kWarningThrottleNs) { 4961 ALOGW("RecordThread: buffer overflow"); 4962 lastWarning = now; 4963 } 4964 } 4965 // Release the processor for a while before asking for a new buffer. 4966 // This will give the application more chance to read from the buffer and 4967 // clear the overflow. 4968 usleep(kRecordThreadSleepUs); 4969 } 4970 } 4971 // enable changes in effect chain 4972 unlockEffectChains(effectChains); 4973 effectChains.clear(); 4974 } 4975 4976 if (!mStandby) { 4977 mInput->stream->common.standby(&mInput->stream->common); 4978 } 4979 mActiveTrack.clear(); 4980 4981 mStartStopCond.broadcast(); 4982 4983 releaseWakeLock(); 4984 4985 ALOGV("RecordThread %p exiting", this); 4986 return false; 4987} 4988 4989 4990sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4991 const sp<AudioFlinger::Client>& client, 4992 uint32_t sampleRate, 4993 audio_format_t format, 4994 int channelMask, 4995 int frameCount, 4996 int sessionId, 4997 status_t *status) 4998{ 4999 sp<RecordTrack> track; 5000 status_t lStatus; 5001 5002 lStatus = initCheck(); 5003 if (lStatus != NO_ERROR) { 5004 ALOGE("Audio driver not initialized."); 5005 goto Exit; 5006 } 5007 5008 { // scope for mLock 5009 Mutex::Autolock _l(mLock); 5010 5011 track = new RecordTrack(this, client, sampleRate, 5012 format, channelMask, frameCount, sessionId); 5013 5014 if (track->getCblk() == 0) { 5015 lStatus = NO_MEMORY; 5016 goto Exit; 5017 } 5018 5019 mTrack = track.get(); 5020 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5021 bool suspend = audio_is_bluetooth_sco_device( 5022 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5023 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5024 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5025 } 5026 lStatus = NO_ERROR; 5027 5028Exit: 5029 if (status) { 5030 *status = lStatus; 5031 } 5032 return track; 5033} 5034 5035status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5036{ 5037 ALOGV("RecordThread::start tid=%d", tid); 5038 sp <ThreadBase> strongMe = this; 5039 status_t status = NO_ERROR; 5040 { 5041 AutoMutex lock(mLock); 5042 if (mActiveTrack != 0) { 5043 if (recordTrack != mActiveTrack.get()) { 5044 status = -EBUSY; 5045 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5046 mActiveTrack->mState = TrackBase::ACTIVE; 5047 } 5048 return status; 5049 } 5050 5051 recordTrack->mState = TrackBase::IDLE; 5052 mActiveTrack = recordTrack; 5053 mLock.unlock(); 5054 status_t status = AudioSystem::startInput(mId); 5055 mLock.lock(); 5056 if (status != NO_ERROR) { 5057 mActiveTrack.clear(); 5058 return status; 5059 } 5060 mRsmpInIndex = mFrameCount; 5061 mBytesRead = 0; 5062 if (mResampler != NULL) { 5063 mResampler->reset(); 5064 } 5065 mActiveTrack->mState = TrackBase::RESUMING; 5066 // signal thread to start 5067 ALOGV("Signal record thread"); 5068 mWaitWorkCV.signal(); 5069 // do not wait for mStartStopCond if exiting 5070 if (exitPending()) { 5071 mActiveTrack.clear(); 5072 status = INVALID_OPERATION; 5073 goto startError; 5074 } 5075 mStartStopCond.wait(mLock); 5076 if (mActiveTrack == 0) { 5077 ALOGV("Record failed to start"); 5078 status = BAD_VALUE; 5079 goto startError; 5080 } 5081 ALOGV("Record started OK"); 5082 return status; 5083 } 5084startError: 5085 AudioSystem::stopInput(mId); 5086 return status; 5087} 5088 5089void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5090 ALOGV("RecordThread::stop"); 5091 sp <ThreadBase> strongMe = this; 5092 { 5093 AutoMutex lock(mLock); 5094 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5095 mActiveTrack->mState = TrackBase::PAUSING; 5096 // do not wait for mStartStopCond if exiting 5097 if (exitPending()) { 5098 return; 5099 } 5100 mStartStopCond.wait(mLock); 5101 // if we have been restarted, recordTrack == mActiveTrack.get() here 5102 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5103 mLock.unlock(); 5104 AudioSystem::stopInput(mId); 5105 mLock.lock(); 5106 ALOGV("Record stopped OK"); 5107 } 5108 } 5109 } 5110} 5111 5112status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5113{ 5114 const size_t SIZE = 256; 5115 char buffer[SIZE]; 5116 String8 result; 5117 5118 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5119 result.append(buffer); 5120 5121 if (mActiveTrack != 0) { 5122 result.append("Active Track:\n"); 5123 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5124 mActiveTrack->dump(buffer, SIZE); 5125 result.append(buffer); 5126 5127 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5128 result.append(buffer); 5129 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5130 result.append(buffer); 5131 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5132 result.append(buffer); 5133 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5134 result.append(buffer); 5135 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5136 result.append(buffer); 5137 5138 5139 } else { 5140 result.append("No record client\n"); 5141 } 5142 write(fd, result.string(), result.size()); 5143 5144 dumpBase(fd, args); 5145 dumpEffectChains(fd, args); 5146 5147 return NO_ERROR; 5148} 5149 5150status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5151{ 5152 size_t framesReq = buffer->frameCount; 5153 size_t framesReady = mFrameCount - mRsmpInIndex; 5154 int channelCount; 5155 5156 if (framesReady == 0) { 5157 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5158 if (mBytesRead < 0) { 5159 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5160 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5161 // Force input into standby so that it tries to 5162 // recover at next read attempt 5163 mInput->stream->common.standby(&mInput->stream->common); 5164 usleep(kRecordThreadSleepUs); 5165 } 5166 buffer->raw = NULL; 5167 buffer->frameCount = 0; 5168 return NOT_ENOUGH_DATA; 5169 } 5170 mRsmpInIndex = 0; 5171 framesReady = mFrameCount; 5172 } 5173 5174 if (framesReq > framesReady) { 5175 framesReq = framesReady; 5176 } 5177 5178 if (mChannelCount == 1 && mReqChannelCount == 2) { 5179 channelCount = 1; 5180 } else { 5181 channelCount = 2; 5182 } 5183 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5184 buffer->frameCount = framesReq; 5185 return NO_ERROR; 5186} 5187 5188void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5189{ 5190 mRsmpInIndex += buffer->frameCount; 5191 buffer->frameCount = 0; 5192} 5193 5194bool AudioFlinger::RecordThread::checkForNewParameters_l() 5195{ 5196 bool reconfig = false; 5197 5198 while (!mNewParameters.isEmpty()) { 5199 status_t status = NO_ERROR; 5200 String8 keyValuePair = mNewParameters[0]; 5201 AudioParameter param = AudioParameter(keyValuePair); 5202 int value; 5203 audio_format_t reqFormat = mFormat; 5204 int reqSamplingRate = mReqSampleRate; 5205 int reqChannelCount = mReqChannelCount; 5206 5207 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5208 reqSamplingRate = value; 5209 reconfig = true; 5210 } 5211 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5212 reqFormat = (audio_format_t) value; 5213 reconfig = true; 5214 } 5215 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5216 reqChannelCount = popcount(value); 5217 reconfig = true; 5218 } 5219 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5220 // do not accept frame count changes if tracks are open as the track buffer 5221 // size depends on frame count and correct behavior would not be guaranteed 5222 // if frame count is changed after track creation 5223 if (mActiveTrack != 0) { 5224 status = INVALID_OPERATION; 5225 } else { 5226 reconfig = true; 5227 } 5228 } 5229 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5230 // forward device change to effects that have requested to be 5231 // aware of attached audio device. 5232 for (size_t i = 0; i < mEffectChains.size(); i++) { 5233 mEffectChains[i]->setDevice_l(value); 5234 } 5235 // store input device and output device but do not forward output device to audio HAL. 5236 // Note that status is ignored by the caller for output device 5237 // (see AudioFlinger::setParameters() 5238 if (value & AUDIO_DEVICE_OUT_ALL) { 5239 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5240 status = BAD_VALUE; 5241 } else { 5242 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5243 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5244 if (mTrack != NULL) { 5245 bool suspend = audio_is_bluetooth_sco_device( 5246 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5247 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5248 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5249 } 5250 } 5251 mDevice |= (uint32_t)value; 5252 } 5253 if (status == NO_ERROR) { 5254 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5255 if (status == INVALID_OPERATION) { 5256 mInput->stream->common.standby(&mInput->stream->common); 5257 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5258 } 5259 if (reconfig) { 5260 if (status == BAD_VALUE && 5261 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5262 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5263 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5264 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5265 (reqChannelCount < 3)) { 5266 status = NO_ERROR; 5267 } 5268 if (status == NO_ERROR) { 5269 readInputParameters(); 5270 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5271 } 5272 } 5273 } 5274 5275 mNewParameters.removeAt(0); 5276 5277 mParamStatus = status; 5278 mParamCond.signal(); 5279 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5280 // already timed out waiting for the status and will never signal the condition. 5281 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5282 } 5283 return reconfig; 5284} 5285 5286String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5287{ 5288 char *s; 5289 String8 out_s8 = String8(); 5290 5291 Mutex::Autolock _l(mLock); 5292 if (initCheck() != NO_ERROR) { 5293 return out_s8; 5294 } 5295 5296 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5297 out_s8 = String8(s); 5298 free(s); 5299 return out_s8; 5300} 5301 5302void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5303 AudioSystem::OutputDescriptor desc; 5304 void *param2 = NULL; 5305 5306 switch (event) { 5307 case AudioSystem::INPUT_OPENED: 5308 case AudioSystem::INPUT_CONFIG_CHANGED: 5309 desc.channels = mChannelMask; 5310 desc.samplingRate = mSampleRate; 5311 desc.format = mFormat; 5312 desc.frameCount = mFrameCount; 5313 desc.latency = 0; 5314 param2 = &desc; 5315 break; 5316 5317 case AudioSystem::INPUT_CLOSED: 5318 default: 5319 break; 5320 } 5321 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5322} 5323 5324void AudioFlinger::RecordThread::readInputParameters() 5325{ 5326 delete mRsmpInBuffer; 5327 // mRsmpInBuffer is always assigned a new[] below 5328 delete mRsmpOutBuffer; 5329 mRsmpOutBuffer = NULL; 5330 delete mResampler; 5331 mResampler = NULL; 5332 5333 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5334 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5335 mChannelCount = (uint16_t)popcount(mChannelMask); 5336 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5337 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5338 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5339 mFrameCount = mInputBytes / mFrameSize; 5340 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5341 5342 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5343 { 5344 int channelCount; 5345 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5346 // stereo to mono post process as the resampler always outputs stereo. 5347 if (mChannelCount == 1 && mReqChannelCount == 2) { 5348 channelCount = 1; 5349 } else { 5350 channelCount = 2; 5351 } 5352 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5353 mResampler->setSampleRate(mSampleRate); 5354 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5355 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5356 5357 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5358 if (mChannelCount == 1 && mReqChannelCount == 1) { 5359 mFrameCount >>= 1; 5360 } 5361 5362 } 5363 mRsmpInIndex = mFrameCount; 5364} 5365 5366unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5367{ 5368 Mutex::Autolock _l(mLock); 5369 if (initCheck() != NO_ERROR) { 5370 return 0; 5371 } 5372 5373 return mInput->stream->get_input_frames_lost(mInput->stream); 5374} 5375 5376uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5377{ 5378 Mutex::Autolock _l(mLock); 5379 uint32_t result = 0; 5380 if (getEffectChain_l(sessionId) != 0) { 5381 result = EFFECT_SESSION; 5382 } 5383 5384 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5385 result |= TRACK_SESSION; 5386 } 5387 5388 return result; 5389} 5390 5391AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5392{ 5393 Mutex::Autolock _l(mLock); 5394 return mTrack; 5395} 5396 5397AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5398{ 5399 Mutex::Autolock _l(mLock); 5400 return mInput; 5401} 5402 5403AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5404{ 5405 Mutex::Autolock _l(mLock); 5406 AudioStreamIn *input = mInput; 5407 mInput = NULL; 5408 return input; 5409} 5410 5411// this method must always be called either with ThreadBase mLock held or inside the thread loop 5412audio_stream_t* AudioFlinger::RecordThread::stream() 5413{ 5414 if (mInput == NULL) { 5415 return NULL; 5416 } 5417 return &mInput->stream->common; 5418} 5419 5420 5421// ---------------------------------------------------------------------------- 5422 5423audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5424 uint32_t *pSamplingRate, 5425 audio_format_t *pFormat, 5426 uint32_t *pChannels, 5427 uint32_t *pLatencyMs, 5428 uint32_t flags) 5429{ 5430 status_t status; 5431 PlaybackThread *thread = NULL; 5432 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5433 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5434 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5435 uint32_t channels = pChannels ? *pChannels : 0; 5436 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5437 audio_stream_out_t *outStream; 5438 audio_hw_device_t *outHwDev; 5439 5440 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5441 pDevices ? *pDevices : 0, 5442 samplingRate, 5443 format, 5444 channels, 5445 flags); 5446 5447 if (pDevices == NULL || *pDevices == 0) { 5448 return 0; 5449 } 5450 5451 Mutex::Autolock _l(mLock); 5452 5453 outHwDev = findSuitableHwDev_l(*pDevices); 5454 if (outHwDev == NULL) 5455 return 0; 5456 5457 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5458 &channels, &samplingRate, &outStream); 5459 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5460 outStream, 5461 samplingRate, 5462 format, 5463 channels, 5464 status); 5465 5466 mHardwareStatus = AUDIO_HW_IDLE; 5467 if (outStream != NULL) { 5468 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5469 audio_io_handle_t id = nextUniqueId(); 5470 5471 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5472 (format != AUDIO_FORMAT_PCM_16_BIT) || 5473 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5474 thread = new DirectOutputThread(this, output, id, *pDevices); 5475 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5476 } else { 5477 thread = new MixerThread(this, output, id, *pDevices); 5478 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5479 } 5480 mPlaybackThreads.add(id, thread); 5481 5482 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5483 if (pFormat != NULL) *pFormat = format; 5484 if (pChannels != NULL) *pChannels = channels; 5485 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5486 5487 // notify client processes of the new output creation 5488 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5489 return id; 5490 } 5491 5492 return 0; 5493} 5494 5495audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5496 audio_io_handle_t output2) 5497{ 5498 Mutex::Autolock _l(mLock); 5499 MixerThread *thread1 = checkMixerThread_l(output1); 5500 MixerThread *thread2 = checkMixerThread_l(output2); 5501 5502 if (thread1 == NULL || thread2 == NULL) { 5503 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5504 return 0; 5505 } 5506 5507 audio_io_handle_t id = nextUniqueId(); 5508 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5509 thread->addOutputTrack(thread2); 5510 mPlaybackThreads.add(id, thread); 5511 // notify client processes of the new output creation 5512 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5513 return id; 5514} 5515 5516status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5517{ 5518 // keep strong reference on the playback thread so that 5519 // it is not destroyed while exit() is executed 5520 sp <PlaybackThread> thread; 5521 { 5522 Mutex::Autolock _l(mLock); 5523 thread = checkPlaybackThread_l(output); 5524 if (thread == NULL) { 5525 return BAD_VALUE; 5526 } 5527 5528 ALOGV("closeOutput() %d", output); 5529 5530 if (thread->type() == ThreadBase::MIXER) { 5531 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5532 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5533 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5534 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5535 } 5536 } 5537 } 5538 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5539 mPlaybackThreads.removeItem(output); 5540 } 5541 thread->exit(); 5542 // The thread entity (active unit of execution) is no longer running here, 5543 // but the ThreadBase container still exists. 5544 5545 if (thread->type() != ThreadBase::DUPLICATING) { 5546 AudioStreamOut *out = thread->clearOutput(); 5547 assert(out != NULL); 5548 // from now on thread->mOutput is NULL 5549 out->hwDev->close_output_stream(out->hwDev, out->stream); 5550 delete out; 5551 } 5552 return NO_ERROR; 5553} 5554 5555status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5556{ 5557 Mutex::Autolock _l(mLock); 5558 PlaybackThread *thread = checkPlaybackThread_l(output); 5559 5560 if (thread == NULL) { 5561 return BAD_VALUE; 5562 } 5563 5564 ALOGV("suspendOutput() %d", output); 5565 thread->suspend(); 5566 5567 return NO_ERROR; 5568} 5569 5570status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5571{ 5572 Mutex::Autolock _l(mLock); 5573 PlaybackThread *thread = checkPlaybackThread_l(output); 5574 5575 if (thread == NULL) { 5576 return BAD_VALUE; 5577 } 5578 5579 ALOGV("restoreOutput() %d", output); 5580 5581 thread->restore(); 5582 5583 return NO_ERROR; 5584} 5585 5586audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5587 uint32_t *pSamplingRate, 5588 audio_format_t *pFormat, 5589 uint32_t *pChannels, 5590 audio_in_acoustics_t acoustics) 5591{ 5592 status_t status; 5593 RecordThread *thread = NULL; 5594 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5595 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5596 uint32_t channels = pChannels ? *pChannels : 0; 5597 uint32_t reqSamplingRate = samplingRate; 5598 audio_format_t reqFormat = format; 5599 uint32_t reqChannels = channels; 5600 audio_stream_in_t *inStream; 5601 audio_hw_device_t *inHwDev; 5602 5603 if (pDevices == NULL || *pDevices == 0) { 5604 return 0; 5605 } 5606 5607 Mutex::Autolock _l(mLock); 5608 5609 inHwDev = findSuitableHwDev_l(*pDevices); 5610 if (inHwDev == NULL) 5611 return 0; 5612 5613 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5614 &channels, &samplingRate, 5615 acoustics, 5616 &inStream); 5617 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5618 inStream, 5619 samplingRate, 5620 format, 5621 channels, 5622 acoustics, 5623 status); 5624 5625 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5626 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5627 // or stereo to mono conversions on 16 bit PCM inputs. 5628 if (inStream == NULL && status == BAD_VALUE && 5629 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5630 (samplingRate <= 2 * reqSamplingRate) && 5631 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5632 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5633 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5634 &channels, &samplingRate, 5635 acoustics, 5636 &inStream); 5637 } 5638 5639 if (inStream != NULL) { 5640 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5641 5642 audio_io_handle_t id = nextUniqueId(); 5643 // Start record thread 5644 // RecorThread require both input and output device indication to forward to audio 5645 // pre processing modules 5646 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5647 thread = new RecordThread(this, 5648 input, 5649 reqSamplingRate, 5650 reqChannels, 5651 id, 5652 device); 5653 mRecordThreads.add(id, thread); 5654 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5655 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5656 if (pFormat != NULL) *pFormat = format; 5657 if (pChannels != NULL) *pChannels = reqChannels; 5658 5659 input->stream->common.standby(&input->stream->common); 5660 5661 // notify client processes of the new input creation 5662 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5663 return id; 5664 } 5665 5666 return 0; 5667} 5668 5669status_t AudioFlinger::closeInput(audio_io_handle_t input) 5670{ 5671 // keep strong reference on the record thread so that 5672 // it is not destroyed while exit() is executed 5673 sp <RecordThread> thread; 5674 { 5675 Mutex::Autolock _l(mLock); 5676 thread = checkRecordThread_l(input); 5677 if (thread == NULL) { 5678 return BAD_VALUE; 5679 } 5680 5681 ALOGV("closeInput() %d", input); 5682 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5683 mRecordThreads.removeItem(input); 5684 } 5685 thread->exit(); 5686 // The thread entity (active unit of execution) is no longer running here, 5687 // but the ThreadBase container still exists. 5688 5689 AudioStreamIn *in = thread->clearInput(); 5690 assert(in != NULL); 5691 // from now on thread->mInput is NULL 5692 in->hwDev->close_input_stream(in->hwDev, in->stream); 5693 delete in; 5694 5695 return NO_ERROR; 5696} 5697 5698status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5699{ 5700 Mutex::Autolock _l(mLock); 5701 MixerThread *dstThread = checkMixerThread_l(output); 5702 if (dstThread == NULL) { 5703 ALOGW("setStreamOutput() bad output id %d", output); 5704 return BAD_VALUE; 5705 } 5706 5707 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5708 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5709 5710 dstThread->setStreamValid(stream, true); 5711 5712 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5713 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5714 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5715 MixerThread *srcThread = (MixerThread *)thread; 5716 srcThread->setStreamValid(stream, false); 5717 srcThread->invalidateTracks(stream); 5718 } 5719 } 5720 5721 return NO_ERROR; 5722} 5723 5724 5725int AudioFlinger::newAudioSessionId() 5726{ 5727 return nextUniqueId(); 5728} 5729 5730void AudioFlinger::acquireAudioSessionId(int audioSession) 5731{ 5732 Mutex::Autolock _l(mLock); 5733 pid_t caller = IPCThreadState::self()->getCallingPid(); 5734 ALOGV("acquiring %d from %d", audioSession, caller); 5735 size_t num = mAudioSessionRefs.size(); 5736 for (size_t i = 0; i< num; i++) { 5737 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5738 if (ref->sessionid == audioSession && ref->pid == caller) { 5739 ref->cnt++; 5740 ALOGV(" incremented refcount to %d", ref->cnt); 5741 return; 5742 } 5743 } 5744 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5745 ALOGV(" added new entry for %d", audioSession); 5746} 5747 5748void AudioFlinger::releaseAudioSessionId(int audioSession) 5749{ 5750 Mutex::Autolock _l(mLock); 5751 pid_t caller = IPCThreadState::self()->getCallingPid(); 5752 ALOGV("releasing %d from %d", audioSession, caller); 5753 size_t num = mAudioSessionRefs.size(); 5754 for (size_t i = 0; i< num; i++) { 5755 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5756 if (ref->sessionid == audioSession && ref->pid == caller) { 5757 ref->cnt--; 5758 ALOGV(" decremented refcount to %d", ref->cnt); 5759 if (ref->cnt == 0) { 5760 mAudioSessionRefs.removeAt(i); 5761 delete ref; 5762 purgeStaleEffects_l(); 5763 } 5764 return; 5765 } 5766 } 5767 ALOGW("session id %d not found for pid %d", audioSession, caller); 5768} 5769 5770void AudioFlinger::purgeStaleEffects_l() { 5771 5772 ALOGV("purging stale effects"); 5773 5774 Vector< sp<EffectChain> > chains; 5775 5776 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5777 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5778 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5779 sp<EffectChain> ec = t->mEffectChains[j]; 5780 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5781 chains.push(ec); 5782 } 5783 } 5784 } 5785 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5786 sp<RecordThread> t = mRecordThreads.valueAt(i); 5787 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5788 sp<EffectChain> ec = t->mEffectChains[j]; 5789 chains.push(ec); 5790 } 5791 } 5792 5793 for (size_t i = 0; i < chains.size(); i++) { 5794 sp<EffectChain> ec = chains[i]; 5795 int sessionid = ec->sessionId(); 5796 sp<ThreadBase> t = ec->mThread.promote(); 5797 if (t == 0) { 5798 continue; 5799 } 5800 size_t numsessionrefs = mAudioSessionRefs.size(); 5801 bool found = false; 5802 for (size_t k = 0; k < numsessionrefs; k++) { 5803 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5804 if (ref->sessionid == sessionid) { 5805 ALOGV(" session %d still exists for %d with %d refs", 5806 sessionid, ref->pid, ref->cnt); 5807 found = true; 5808 break; 5809 } 5810 } 5811 if (!found) { 5812 // remove all effects from the chain 5813 while (ec->mEffects.size()) { 5814 sp<EffectModule> effect = ec->mEffects[0]; 5815 effect->unPin(); 5816 Mutex::Autolock _l (t->mLock); 5817 t->removeEffect_l(effect); 5818 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5819 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5820 if (handle != 0) { 5821 handle->mEffect.clear(); 5822 if (handle->mHasControl && handle->mEnabled) { 5823 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5824 } 5825 } 5826 } 5827 AudioSystem::unregisterEffect(effect->id()); 5828 } 5829 } 5830 } 5831 return; 5832} 5833 5834// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5835AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5836{ 5837 return mPlaybackThreads.valueFor(output).get(); 5838} 5839 5840// checkMixerThread_l() must be called with AudioFlinger::mLock held 5841AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5842{ 5843 PlaybackThread *thread = checkPlaybackThread_l(output); 5844 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5845} 5846 5847// checkRecordThread_l() must be called with AudioFlinger::mLock held 5848AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5849{ 5850 return mRecordThreads.valueFor(input).get(); 5851} 5852 5853uint32_t AudioFlinger::nextUniqueId() 5854{ 5855 return android_atomic_inc(&mNextUniqueId); 5856} 5857 5858AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5859{ 5860 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5861 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5862 AudioStreamOut *output = thread->getOutput(); 5863 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5864 return thread; 5865 } 5866 } 5867 return NULL; 5868} 5869 5870uint32_t AudioFlinger::primaryOutputDevice_l() 5871{ 5872 PlaybackThread *thread = primaryPlaybackThread_l(); 5873 5874 if (thread == NULL) { 5875 return 0; 5876 } 5877 5878 return thread->device(); 5879} 5880 5881 5882// ---------------------------------------------------------------------------- 5883// Effect management 5884// ---------------------------------------------------------------------------- 5885 5886 5887status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5888{ 5889 Mutex::Autolock _l(mLock); 5890 return EffectQueryNumberEffects(numEffects); 5891} 5892 5893status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5894{ 5895 Mutex::Autolock _l(mLock); 5896 return EffectQueryEffect(index, descriptor); 5897} 5898 5899status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5900 effect_descriptor_t *descriptor) const 5901{ 5902 Mutex::Autolock _l(mLock); 5903 return EffectGetDescriptor(pUuid, descriptor); 5904} 5905 5906 5907sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5908 effect_descriptor_t *pDesc, 5909 const sp<IEffectClient>& effectClient, 5910 int32_t priority, 5911 audio_io_handle_t io, 5912 int sessionId, 5913 status_t *status, 5914 int *id, 5915 int *enabled) 5916{ 5917 status_t lStatus = NO_ERROR; 5918 sp<EffectHandle> handle; 5919 effect_descriptor_t desc; 5920 5921 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5922 pid, effectClient.get(), priority, sessionId, io); 5923 5924 if (pDesc == NULL) { 5925 lStatus = BAD_VALUE; 5926 goto Exit; 5927 } 5928 5929 // check audio settings permission for global effects 5930 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5931 lStatus = PERMISSION_DENIED; 5932 goto Exit; 5933 } 5934 5935 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5936 // that can only be created by audio policy manager (running in same process) 5937 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5938 lStatus = PERMISSION_DENIED; 5939 goto Exit; 5940 } 5941 5942 if (io == 0) { 5943 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5944 // output must be specified by AudioPolicyManager when using session 5945 // AUDIO_SESSION_OUTPUT_STAGE 5946 lStatus = BAD_VALUE; 5947 goto Exit; 5948 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5949 // if the output returned by getOutputForEffect() is removed before we lock the 5950 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5951 // and we will exit safely 5952 io = AudioSystem::getOutputForEffect(&desc); 5953 } 5954 } 5955 5956 { 5957 Mutex::Autolock _l(mLock); 5958 5959 5960 if (!EffectIsNullUuid(&pDesc->uuid)) { 5961 // if uuid is specified, request effect descriptor 5962 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5963 if (lStatus < 0) { 5964 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5965 goto Exit; 5966 } 5967 } else { 5968 // if uuid is not specified, look for an available implementation 5969 // of the required type in effect factory 5970 if (EffectIsNullUuid(&pDesc->type)) { 5971 ALOGW("createEffect() no effect type"); 5972 lStatus = BAD_VALUE; 5973 goto Exit; 5974 } 5975 uint32_t numEffects = 0; 5976 effect_descriptor_t d; 5977 d.flags = 0; // prevent compiler warning 5978 bool found = false; 5979 5980 lStatus = EffectQueryNumberEffects(&numEffects); 5981 if (lStatus < 0) { 5982 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5983 goto Exit; 5984 } 5985 for (uint32_t i = 0; i < numEffects; i++) { 5986 lStatus = EffectQueryEffect(i, &desc); 5987 if (lStatus < 0) { 5988 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5989 continue; 5990 } 5991 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5992 // If matching type found save effect descriptor. If the session is 5993 // 0 and the effect is not auxiliary, continue enumeration in case 5994 // an auxiliary version of this effect type is available 5995 found = true; 5996 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5997 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5998 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5999 break; 6000 } 6001 } 6002 } 6003 if (!found) { 6004 lStatus = BAD_VALUE; 6005 ALOGW("createEffect() effect not found"); 6006 goto Exit; 6007 } 6008 // For same effect type, chose auxiliary version over insert version if 6009 // connect to output mix (Compliance to OpenSL ES) 6010 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6011 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6012 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6013 } 6014 } 6015 6016 // Do not allow auxiliary effects on a session different from 0 (output mix) 6017 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6018 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6019 lStatus = INVALID_OPERATION; 6020 goto Exit; 6021 } 6022 6023 // check recording permission for visualizer 6024 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6025 !recordingAllowed()) { 6026 lStatus = PERMISSION_DENIED; 6027 goto Exit; 6028 } 6029 6030 // return effect descriptor 6031 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6032 6033 // If output is not specified try to find a matching audio session ID in one of the 6034 // output threads. 6035 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6036 // because of code checking output when entering the function. 6037 // Note: io is never 0 when creating an effect on an input 6038 if (io == 0) { 6039 // look for the thread where the specified audio session is present 6040 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6041 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6042 io = mPlaybackThreads.keyAt(i); 6043 break; 6044 } 6045 } 6046 if (io == 0) { 6047 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6048 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6049 io = mRecordThreads.keyAt(i); 6050 break; 6051 } 6052 } 6053 } 6054 // If no output thread contains the requested session ID, default to 6055 // first output. The effect chain will be moved to the correct output 6056 // thread when a track with the same session ID is created 6057 if (io == 0 && mPlaybackThreads.size()) { 6058 io = mPlaybackThreads.keyAt(0); 6059 } 6060 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6061 } 6062 ThreadBase *thread = checkRecordThread_l(io); 6063 if (thread == NULL) { 6064 thread = checkPlaybackThread_l(io); 6065 if (thread == NULL) { 6066 ALOGE("createEffect() unknown output thread"); 6067 lStatus = BAD_VALUE; 6068 goto Exit; 6069 } 6070 } 6071 6072 sp<Client> client = registerPid_l(pid); 6073 6074 // create effect on selected output thread 6075 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6076 &desc, enabled, &lStatus); 6077 if (handle != 0 && id != NULL) { 6078 *id = handle->id(); 6079 } 6080 } 6081 6082Exit: 6083 if(status) { 6084 *status = lStatus; 6085 } 6086 return handle; 6087} 6088 6089status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6090 audio_io_handle_t dstOutput) 6091{ 6092 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6093 sessionId, srcOutput, dstOutput); 6094 Mutex::Autolock _l(mLock); 6095 if (srcOutput == dstOutput) { 6096 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6097 return NO_ERROR; 6098 } 6099 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6100 if (srcThread == NULL) { 6101 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6102 return BAD_VALUE; 6103 } 6104 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6105 if (dstThread == NULL) { 6106 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6107 return BAD_VALUE; 6108 } 6109 6110 Mutex::Autolock _dl(dstThread->mLock); 6111 Mutex::Autolock _sl(srcThread->mLock); 6112 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6113 6114 return NO_ERROR; 6115} 6116 6117// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6118status_t AudioFlinger::moveEffectChain_l(int sessionId, 6119 AudioFlinger::PlaybackThread *srcThread, 6120 AudioFlinger::PlaybackThread *dstThread, 6121 bool reRegister) 6122{ 6123 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6124 sessionId, srcThread, dstThread); 6125 6126 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6127 if (chain == 0) { 6128 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6129 sessionId, srcThread); 6130 return INVALID_OPERATION; 6131 } 6132 6133 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6134 // so that a new chain is created with correct parameters when first effect is added. This is 6135 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6136 // removed. 6137 srcThread->removeEffectChain_l(chain); 6138 6139 // transfer all effects one by one so that new effect chain is created on new thread with 6140 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6141 audio_io_handle_t dstOutput = dstThread->id(); 6142 sp<EffectChain> dstChain; 6143 uint32_t strategy = 0; // prevent compiler warning 6144 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6145 while (effect != 0) { 6146 srcThread->removeEffect_l(effect); 6147 dstThread->addEffect_l(effect); 6148 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6149 if (effect->state() == EffectModule::ACTIVE || 6150 effect->state() == EffectModule::STOPPING) { 6151 effect->start(); 6152 } 6153 // if the move request is not received from audio policy manager, the effect must be 6154 // re-registered with the new strategy and output 6155 if (dstChain == 0) { 6156 dstChain = effect->chain().promote(); 6157 if (dstChain == 0) { 6158 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6159 srcThread->addEffect_l(effect); 6160 return NO_INIT; 6161 } 6162 strategy = dstChain->strategy(); 6163 } 6164 if (reRegister) { 6165 AudioSystem::unregisterEffect(effect->id()); 6166 AudioSystem::registerEffect(&effect->desc(), 6167 dstOutput, 6168 strategy, 6169 sessionId, 6170 effect->id()); 6171 } 6172 effect = chain->getEffectFromId_l(0); 6173 } 6174 6175 return NO_ERROR; 6176} 6177 6178 6179// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6180sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6181 const sp<AudioFlinger::Client>& client, 6182 const sp<IEffectClient>& effectClient, 6183 int32_t priority, 6184 int sessionId, 6185 effect_descriptor_t *desc, 6186 int *enabled, 6187 status_t *status 6188 ) 6189{ 6190 sp<EffectModule> effect; 6191 sp<EffectHandle> handle; 6192 status_t lStatus; 6193 sp<EffectChain> chain; 6194 bool chainCreated = false; 6195 bool effectCreated = false; 6196 bool effectRegistered = false; 6197 6198 lStatus = initCheck(); 6199 if (lStatus != NO_ERROR) { 6200 ALOGW("createEffect_l() Audio driver not initialized."); 6201 goto Exit; 6202 } 6203 6204 // Do not allow effects with session ID 0 on direct output or duplicating threads 6205 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6206 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6207 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6208 desc->name, sessionId); 6209 lStatus = BAD_VALUE; 6210 goto Exit; 6211 } 6212 // Only Pre processor effects are allowed on input threads and only on input threads 6213 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6214 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6215 desc->name, desc->flags, mType); 6216 lStatus = BAD_VALUE; 6217 goto Exit; 6218 } 6219 6220 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6221 6222 { // scope for mLock 6223 Mutex::Autolock _l(mLock); 6224 6225 // check for existing effect chain with the requested audio session 6226 chain = getEffectChain_l(sessionId); 6227 if (chain == 0) { 6228 // create a new chain for this session 6229 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6230 chain = new EffectChain(this, sessionId); 6231 addEffectChain_l(chain); 6232 chain->setStrategy(getStrategyForSession_l(sessionId)); 6233 chainCreated = true; 6234 } else { 6235 effect = chain->getEffectFromDesc_l(desc); 6236 } 6237 6238 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6239 6240 if (effect == 0) { 6241 int id = mAudioFlinger->nextUniqueId(); 6242 // Check CPU and memory usage 6243 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6244 if (lStatus != NO_ERROR) { 6245 goto Exit; 6246 } 6247 effectRegistered = true; 6248 // create a new effect module if none present in the chain 6249 effect = new EffectModule(this, chain, desc, id, sessionId); 6250 lStatus = effect->status(); 6251 if (lStatus != NO_ERROR) { 6252 goto Exit; 6253 } 6254 lStatus = chain->addEffect_l(effect); 6255 if (lStatus != NO_ERROR) { 6256 goto Exit; 6257 } 6258 effectCreated = true; 6259 6260 effect->setDevice(mDevice); 6261 effect->setMode(mAudioFlinger->getMode()); 6262 } 6263 // create effect handle and connect it to effect module 6264 handle = new EffectHandle(effect, client, effectClient, priority); 6265 lStatus = effect->addHandle(handle); 6266 if (enabled != NULL) { 6267 *enabled = (int)effect->isEnabled(); 6268 } 6269 } 6270 6271Exit: 6272 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6273 Mutex::Autolock _l(mLock); 6274 if (effectCreated) { 6275 chain->removeEffect_l(effect); 6276 } 6277 if (effectRegistered) { 6278 AudioSystem::unregisterEffect(effect->id()); 6279 } 6280 if (chainCreated) { 6281 removeEffectChain_l(chain); 6282 } 6283 handle.clear(); 6284 } 6285 6286 if(status) { 6287 *status = lStatus; 6288 } 6289 return handle; 6290} 6291 6292sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6293{ 6294 sp<EffectChain> chain = getEffectChain_l(sessionId); 6295 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6296} 6297 6298// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6299// PlaybackThread::mLock held 6300status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6301{ 6302 // check for existing effect chain with the requested audio session 6303 int sessionId = effect->sessionId(); 6304 sp<EffectChain> chain = getEffectChain_l(sessionId); 6305 bool chainCreated = false; 6306 6307 if (chain == 0) { 6308 // create a new chain for this session 6309 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6310 chain = new EffectChain(this, sessionId); 6311 addEffectChain_l(chain); 6312 chain->setStrategy(getStrategyForSession_l(sessionId)); 6313 chainCreated = true; 6314 } 6315 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6316 6317 if (chain->getEffectFromId_l(effect->id()) != 0) { 6318 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6319 this, effect->desc().name, chain.get()); 6320 return BAD_VALUE; 6321 } 6322 6323 status_t status = chain->addEffect_l(effect); 6324 if (status != NO_ERROR) { 6325 if (chainCreated) { 6326 removeEffectChain_l(chain); 6327 } 6328 return status; 6329 } 6330 6331 effect->setDevice(mDevice); 6332 effect->setMode(mAudioFlinger->getMode()); 6333 return NO_ERROR; 6334} 6335 6336void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6337 6338 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6339 effect_descriptor_t desc = effect->desc(); 6340 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6341 detachAuxEffect_l(effect->id()); 6342 } 6343 6344 sp<EffectChain> chain = effect->chain().promote(); 6345 if (chain != 0) { 6346 // remove effect chain if removing last effect 6347 if (chain->removeEffect_l(effect) == 0) { 6348 removeEffectChain_l(chain); 6349 } 6350 } else { 6351 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6352 } 6353} 6354 6355void AudioFlinger::ThreadBase::lockEffectChains_l( 6356 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6357{ 6358 effectChains = mEffectChains; 6359 for (size_t i = 0; i < mEffectChains.size(); i++) { 6360 mEffectChains[i]->lock(); 6361 } 6362} 6363 6364void AudioFlinger::ThreadBase::unlockEffectChains( 6365 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6366{ 6367 for (size_t i = 0; i < effectChains.size(); i++) { 6368 effectChains[i]->unlock(); 6369 } 6370} 6371 6372sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6373{ 6374 Mutex::Autolock _l(mLock); 6375 return getEffectChain_l(sessionId); 6376} 6377 6378sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6379{ 6380 size_t size = mEffectChains.size(); 6381 for (size_t i = 0; i < size; i++) { 6382 if (mEffectChains[i]->sessionId() == sessionId) { 6383 return mEffectChains[i]; 6384 } 6385 } 6386 return 0; 6387} 6388 6389void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6390{ 6391 Mutex::Autolock _l(mLock); 6392 size_t size = mEffectChains.size(); 6393 for (size_t i = 0; i < size; i++) { 6394 mEffectChains[i]->setMode_l(mode); 6395 } 6396} 6397 6398void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6399 const wp<EffectHandle>& handle, 6400 bool unpinIfLast) { 6401 6402 Mutex::Autolock _l(mLock); 6403 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6404 // delete the effect module if removing last handle on it 6405 if (effect->removeHandle(handle) == 0) { 6406 if (!effect->isPinned() || unpinIfLast) { 6407 removeEffect_l(effect); 6408 AudioSystem::unregisterEffect(effect->id()); 6409 } 6410 } 6411} 6412 6413status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6414{ 6415 int session = chain->sessionId(); 6416 int16_t *buffer = mMixBuffer; 6417 bool ownsBuffer = false; 6418 6419 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6420 if (session > 0) { 6421 // Only one effect chain can be present in direct output thread and it uses 6422 // the mix buffer as input 6423 if (mType != DIRECT) { 6424 size_t numSamples = mFrameCount * mChannelCount; 6425 buffer = new int16_t[numSamples]; 6426 memset(buffer, 0, numSamples * sizeof(int16_t)); 6427 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6428 ownsBuffer = true; 6429 } 6430 6431 // Attach all tracks with same session ID to this chain. 6432 for (size_t i = 0; i < mTracks.size(); ++i) { 6433 sp<Track> track = mTracks[i]; 6434 if (session == track->sessionId()) { 6435 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6436 track->setMainBuffer(buffer); 6437 chain->incTrackCnt(); 6438 } 6439 } 6440 6441 // indicate all active tracks in the chain 6442 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6443 sp<Track> track = mActiveTracks[i].promote(); 6444 if (track == 0) continue; 6445 if (session == track->sessionId()) { 6446 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6447 chain->incActiveTrackCnt(); 6448 } 6449 } 6450 } 6451 6452 chain->setInBuffer(buffer, ownsBuffer); 6453 chain->setOutBuffer(mMixBuffer); 6454 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6455 // chains list in order to be processed last as it contains output stage effects 6456 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6457 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6458 // after track specific effects and before output stage 6459 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6460 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6461 // Effect chain for other sessions are inserted at beginning of effect 6462 // chains list to be processed before output mix effects. Relative order between other 6463 // sessions is not important 6464 size_t size = mEffectChains.size(); 6465 size_t i = 0; 6466 for (i = 0; i < size; i++) { 6467 if (mEffectChains[i]->sessionId() < session) break; 6468 } 6469 mEffectChains.insertAt(chain, i); 6470 checkSuspendOnAddEffectChain_l(chain); 6471 6472 return NO_ERROR; 6473} 6474 6475size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6476{ 6477 int session = chain->sessionId(); 6478 6479 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6480 6481 for (size_t i = 0; i < mEffectChains.size(); i++) { 6482 if (chain == mEffectChains[i]) { 6483 mEffectChains.removeAt(i); 6484 // detach all active tracks from the chain 6485 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6486 sp<Track> track = mActiveTracks[i].promote(); 6487 if (track == 0) continue; 6488 if (session == track->sessionId()) { 6489 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6490 chain.get(), session); 6491 chain->decActiveTrackCnt(); 6492 } 6493 } 6494 6495 // detach all tracks with same session ID from this chain 6496 for (size_t i = 0; i < mTracks.size(); ++i) { 6497 sp<Track> track = mTracks[i]; 6498 if (session == track->sessionId()) { 6499 track->setMainBuffer(mMixBuffer); 6500 chain->decTrackCnt(); 6501 } 6502 } 6503 break; 6504 } 6505 } 6506 return mEffectChains.size(); 6507} 6508 6509status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6510 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6511{ 6512 Mutex::Autolock _l(mLock); 6513 return attachAuxEffect_l(track, EffectId); 6514} 6515 6516status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6517 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6518{ 6519 status_t status = NO_ERROR; 6520 6521 if (EffectId == 0) { 6522 track->setAuxBuffer(0, NULL); 6523 } else { 6524 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6525 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6526 if (effect != 0) { 6527 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6528 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6529 } else { 6530 status = INVALID_OPERATION; 6531 } 6532 } else { 6533 status = BAD_VALUE; 6534 } 6535 } 6536 return status; 6537} 6538 6539void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6540{ 6541 for (size_t i = 0; i < mTracks.size(); ++i) { 6542 sp<Track> track = mTracks[i]; 6543 if (track->auxEffectId() == effectId) { 6544 attachAuxEffect_l(track, 0); 6545 } 6546 } 6547} 6548 6549status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6550{ 6551 // only one chain per input thread 6552 if (mEffectChains.size() != 0) { 6553 return INVALID_OPERATION; 6554 } 6555 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6556 6557 chain->setInBuffer(NULL); 6558 chain->setOutBuffer(NULL); 6559 6560 checkSuspendOnAddEffectChain_l(chain); 6561 6562 mEffectChains.add(chain); 6563 6564 return NO_ERROR; 6565} 6566 6567size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6568{ 6569 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6570 ALOGW_IF(mEffectChains.size() != 1, 6571 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6572 chain.get(), mEffectChains.size(), this); 6573 if (mEffectChains.size() == 1) { 6574 mEffectChains.removeAt(0); 6575 } 6576 return 0; 6577} 6578 6579// ---------------------------------------------------------------------------- 6580// EffectModule implementation 6581// ---------------------------------------------------------------------------- 6582 6583#undef LOG_TAG 6584#define LOG_TAG "AudioFlinger::EffectModule" 6585 6586AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6587 const wp<AudioFlinger::EffectChain>& chain, 6588 effect_descriptor_t *desc, 6589 int id, 6590 int sessionId) 6591 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6592 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6593{ 6594 ALOGV("Constructor %p", this); 6595 int lStatus; 6596 if (thread == NULL) { 6597 return; 6598 } 6599 6600 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6601 6602 // create effect engine from effect factory 6603 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6604 6605 if (mStatus != NO_ERROR) { 6606 return; 6607 } 6608 lStatus = init(); 6609 if (lStatus < 0) { 6610 mStatus = lStatus; 6611 goto Error; 6612 } 6613 6614 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6615 mPinned = true; 6616 } 6617 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6618 return; 6619Error: 6620 EffectRelease(mEffectInterface); 6621 mEffectInterface = NULL; 6622 ALOGV("Constructor Error %d", mStatus); 6623} 6624 6625AudioFlinger::EffectModule::~EffectModule() 6626{ 6627 ALOGV("Destructor %p", this); 6628 if (mEffectInterface != NULL) { 6629 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6630 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6631 sp<ThreadBase> thread = mThread.promote(); 6632 if (thread != 0) { 6633 audio_stream_t *stream = thread->stream(); 6634 if (stream != NULL) { 6635 stream->remove_audio_effect(stream, mEffectInterface); 6636 } 6637 } 6638 } 6639 // release effect engine 6640 EffectRelease(mEffectInterface); 6641 } 6642} 6643 6644status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6645{ 6646 status_t status; 6647 6648 Mutex::Autolock _l(mLock); 6649 int priority = handle->priority(); 6650 size_t size = mHandles.size(); 6651 sp<EffectHandle> h; 6652 size_t i; 6653 for (i = 0; i < size; i++) { 6654 h = mHandles[i].promote(); 6655 if (h == 0) continue; 6656 if (h->priority() <= priority) break; 6657 } 6658 // if inserted in first place, move effect control from previous owner to this handle 6659 if (i == 0) { 6660 bool enabled = false; 6661 if (h != 0) { 6662 enabled = h->enabled(); 6663 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6664 } 6665 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6666 status = NO_ERROR; 6667 } else { 6668 status = ALREADY_EXISTS; 6669 } 6670 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6671 mHandles.insertAt(handle, i); 6672 return status; 6673} 6674 6675size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6676{ 6677 Mutex::Autolock _l(mLock); 6678 size_t size = mHandles.size(); 6679 size_t i; 6680 for (i = 0; i < size; i++) { 6681 if (mHandles[i] == handle) break; 6682 } 6683 if (i == size) { 6684 return size; 6685 } 6686 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6687 6688 bool enabled = false; 6689 EffectHandle *hdl = handle.unsafe_get(); 6690 if (hdl != NULL) { 6691 ALOGV("removeHandle() unsafe_get OK"); 6692 enabled = hdl->enabled(); 6693 } 6694 mHandles.removeAt(i); 6695 size = mHandles.size(); 6696 // if removed from first place, move effect control from this handle to next in line 6697 if (i == 0 && size != 0) { 6698 sp<EffectHandle> h = mHandles[0].promote(); 6699 if (h != 0) { 6700 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6701 } 6702 } 6703 6704 // Prevent calls to process() and other functions on effect interface from now on. 6705 // The effect engine will be released by the destructor when the last strong reference on 6706 // this object is released which can happen after next process is called. 6707 if (size == 0 && !mPinned) { 6708 mState = DESTROYED; 6709 } 6710 6711 return size; 6712} 6713 6714sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6715{ 6716 Mutex::Autolock _l(mLock); 6717 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6718} 6719 6720void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6721{ 6722 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6723 // keep a strong reference on this EffectModule to avoid calling the 6724 // destructor before we exit 6725 sp<EffectModule> keep(this); 6726 { 6727 sp<ThreadBase> thread = mThread.promote(); 6728 if (thread != 0) { 6729 thread->disconnectEffect(keep, handle, unpinIfLast); 6730 } 6731 } 6732} 6733 6734void AudioFlinger::EffectModule::updateState() { 6735 Mutex::Autolock _l(mLock); 6736 6737 switch (mState) { 6738 case RESTART: 6739 reset_l(); 6740 // FALL THROUGH 6741 6742 case STARTING: 6743 // clear auxiliary effect input buffer for next accumulation 6744 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6745 memset(mConfig.inputCfg.buffer.raw, 6746 0, 6747 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6748 } 6749 start_l(); 6750 mState = ACTIVE; 6751 break; 6752 case STOPPING: 6753 stop_l(); 6754 mDisableWaitCnt = mMaxDisableWaitCnt; 6755 mState = STOPPED; 6756 break; 6757 case STOPPED: 6758 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6759 // turn off sequence. 6760 if (--mDisableWaitCnt == 0) { 6761 reset_l(); 6762 mState = IDLE; 6763 } 6764 break; 6765 default: //IDLE , ACTIVE, DESTROYED 6766 break; 6767 } 6768} 6769 6770void AudioFlinger::EffectModule::process() 6771{ 6772 Mutex::Autolock _l(mLock); 6773 6774 if (mState == DESTROYED || mEffectInterface == NULL || 6775 mConfig.inputCfg.buffer.raw == NULL || 6776 mConfig.outputCfg.buffer.raw == NULL) { 6777 return; 6778 } 6779 6780 if (isProcessEnabled()) { 6781 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6782 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6783 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6784 mConfig.inputCfg.buffer.s32, 6785 mConfig.inputCfg.buffer.frameCount/2); 6786 } 6787 6788 // do the actual processing in the effect engine 6789 int ret = (*mEffectInterface)->process(mEffectInterface, 6790 &mConfig.inputCfg.buffer, 6791 &mConfig.outputCfg.buffer); 6792 6793 // force transition to IDLE state when engine is ready 6794 if (mState == STOPPED && ret == -ENODATA) { 6795 mDisableWaitCnt = 1; 6796 } 6797 6798 // clear auxiliary effect input buffer for next accumulation 6799 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6800 memset(mConfig.inputCfg.buffer.raw, 0, 6801 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6802 } 6803 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6804 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6805 // If an insert effect is idle and input buffer is different from output buffer, 6806 // accumulate input onto output 6807 sp<EffectChain> chain = mChain.promote(); 6808 if (chain != 0 && chain->activeTrackCnt() != 0) { 6809 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6810 int16_t *in = mConfig.inputCfg.buffer.s16; 6811 int16_t *out = mConfig.outputCfg.buffer.s16; 6812 for (size_t i = 0; i < frameCnt; i++) { 6813 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6814 } 6815 } 6816 } 6817} 6818 6819void AudioFlinger::EffectModule::reset_l() 6820{ 6821 if (mEffectInterface == NULL) { 6822 return; 6823 } 6824 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6825} 6826 6827status_t AudioFlinger::EffectModule::configure() 6828{ 6829 uint32_t channels; 6830 if (mEffectInterface == NULL) { 6831 return NO_INIT; 6832 } 6833 6834 sp<ThreadBase> thread = mThread.promote(); 6835 if (thread == 0) { 6836 return DEAD_OBJECT; 6837 } 6838 6839 // TODO: handle configuration of effects replacing track process 6840 if (thread->channelCount() == 1) { 6841 channels = AUDIO_CHANNEL_OUT_MONO; 6842 } else { 6843 channels = AUDIO_CHANNEL_OUT_STEREO; 6844 } 6845 6846 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6847 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6848 } else { 6849 mConfig.inputCfg.channels = channels; 6850 } 6851 mConfig.outputCfg.channels = channels; 6852 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6853 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6854 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6855 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6856 mConfig.inputCfg.bufferProvider.cookie = NULL; 6857 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6858 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6859 mConfig.outputCfg.bufferProvider.cookie = NULL; 6860 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6861 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6862 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6863 // Insert effect: 6864 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6865 // always overwrites output buffer: input buffer == output buffer 6866 // - in other sessions: 6867 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6868 // other effect: overwrites output buffer: input buffer == output buffer 6869 // Auxiliary effect: 6870 // accumulates in output buffer: input buffer != output buffer 6871 // Therefore: accumulate <=> input buffer != output buffer 6872 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6873 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6874 } else { 6875 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6876 } 6877 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6878 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6879 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6880 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6881 6882 ALOGV("configure() %p thread %p buffer %p framecount %d", 6883 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6884 6885 status_t cmdStatus; 6886 uint32_t size = sizeof(int); 6887 status_t status = (*mEffectInterface)->command(mEffectInterface, 6888 EFFECT_CMD_SET_CONFIG, 6889 sizeof(effect_config_t), 6890 &mConfig, 6891 &size, 6892 &cmdStatus); 6893 if (status == 0) { 6894 status = cmdStatus; 6895 } 6896 6897 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6898 (1000 * mConfig.outputCfg.buffer.frameCount); 6899 6900 return status; 6901} 6902 6903status_t AudioFlinger::EffectModule::init() 6904{ 6905 Mutex::Autolock _l(mLock); 6906 if (mEffectInterface == NULL) { 6907 return NO_INIT; 6908 } 6909 status_t cmdStatus; 6910 uint32_t size = sizeof(status_t); 6911 status_t status = (*mEffectInterface)->command(mEffectInterface, 6912 EFFECT_CMD_INIT, 6913 0, 6914 NULL, 6915 &size, 6916 &cmdStatus); 6917 if (status == 0) { 6918 status = cmdStatus; 6919 } 6920 return status; 6921} 6922 6923status_t AudioFlinger::EffectModule::start() 6924{ 6925 Mutex::Autolock _l(mLock); 6926 return start_l(); 6927} 6928 6929status_t AudioFlinger::EffectModule::start_l() 6930{ 6931 if (mEffectInterface == NULL) { 6932 return NO_INIT; 6933 } 6934 status_t cmdStatus; 6935 uint32_t size = sizeof(status_t); 6936 status_t status = (*mEffectInterface)->command(mEffectInterface, 6937 EFFECT_CMD_ENABLE, 6938 0, 6939 NULL, 6940 &size, 6941 &cmdStatus); 6942 if (status == 0) { 6943 status = cmdStatus; 6944 } 6945 if (status == 0 && 6946 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6947 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6948 sp<ThreadBase> thread = mThread.promote(); 6949 if (thread != 0) { 6950 audio_stream_t *stream = thread->stream(); 6951 if (stream != NULL) { 6952 stream->add_audio_effect(stream, mEffectInterface); 6953 } 6954 } 6955 } 6956 return status; 6957} 6958 6959status_t AudioFlinger::EffectModule::stop() 6960{ 6961 Mutex::Autolock _l(mLock); 6962 return stop_l(); 6963} 6964 6965status_t AudioFlinger::EffectModule::stop_l() 6966{ 6967 if (mEffectInterface == NULL) { 6968 return NO_INIT; 6969 } 6970 status_t cmdStatus; 6971 uint32_t size = sizeof(status_t); 6972 status_t status = (*mEffectInterface)->command(mEffectInterface, 6973 EFFECT_CMD_DISABLE, 6974 0, 6975 NULL, 6976 &size, 6977 &cmdStatus); 6978 if (status == 0) { 6979 status = cmdStatus; 6980 } 6981 if (status == 0 && 6982 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6983 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6984 sp<ThreadBase> thread = mThread.promote(); 6985 if (thread != 0) { 6986 audio_stream_t *stream = thread->stream(); 6987 if (stream != NULL) { 6988 stream->remove_audio_effect(stream, mEffectInterface); 6989 } 6990 } 6991 } 6992 return status; 6993} 6994 6995status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6996 uint32_t cmdSize, 6997 void *pCmdData, 6998 uint32_t *replySize, 6999 void *pReplyData) 7000{ 7001 Mutex::Autolock _l(mLock); 7002// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7003 7004 if (mState == DESTROYED || mEffectInterface == NULL) { 7005 return NO_INIT; 7006 } 7007 status_t status = (*mEffectInterface)->command(mEffectInterface, 7008 cmdCode, 7009 cmdSize, 7010 pCmdData, 7011 replySize, 7012 pReplyData); 7013 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7014 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7015 for (size_t i = 1; i < mHandles.size(); i++) { 7016 sp<EffectHandle> h = mHandles[i].promote(); 7017 if (h != 0) { 7018 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7019 } 7020 } 7021 } 7022 return status; 7023} 7024 7025status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7026{ 7027 7028 Mutex::Autolock _l(mLock); 7029 ALOGV("setEnabled %p enabled %d", this, enabled); 7030 7031 if (enabled != isEnabled()) { 7032 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7033 if (enabled && status != NO_ERROR) { 7034 return status; 7035 } 7036 7037 switch (mState) { 7038 // going from disabled to enabled 7039 case IDLE: 7040 mState = STARTING; 7041 break; 7042 case STOPPED: 7043 mState = RESTART; 7044 break; 7045 case STOPPING: 7046 mState = ACTIVE; 7047 break; 7048 7049 // going from enabled to disabled 7050 case RESTART: 7051 mState = STOPPED; 7052 break; 7053 case STARTING: 7054 mState = IDLE; 7055 break; 7056 case ACTIVE: 7057 mState = STOPPING; 7058 break; 7059 case DESTROYED: 7060 return NO_ERROR; // simply ignore as we are being destroyed 7061 } 7062 for (size_t i = 1; i < mHandles.size(); i++) { 7063 sp<EffectHandle> h = mHandles[i].promote(); 7064 if (h != 0) { 7065 h->setEnabled(enabled); 7066 } 7067 } 7068 } 7069 return NO_ERROR; 7070} 7071 7072bool AudioFlinger::EffectModule::isEnabled() const 7073{ 7074 switch (mState) { 7075 case RESTART: 7076 case STARTING: 7077 case ACTIVE: 7078 return true; 7079 case IDLE: 7080 case STOPPING: 7081 case STOPPED: 7082 case DESTROYED: 7083 default: 7084 return false; 7085 } 7086} 7087 7088bool AudioFlinger::EffectModule::isProcessEnabled() const 7089{ 7090 switch (mState) { 7091 case RESTART: 7092 case ACTIVE: 7093 case STOPPING: 7094 case STOPPED: 7095 return true; 7096 case IDLE: 7097 case STARTING: 7098 case DESTROYED: 7099 default: 7100 return false; 7101 } 7102} 7103 7104status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7105{ 7106 Mutex::Autolock _l(mLock); 7107 status_t status = NO_ERROR; 7108 7109 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7110 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7111 if (isProcessEnabled() && 7112 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7113 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7114 status_t cmdStatus; 7115 uint32_t volume[2]; 7116 uint32_t *pVolume = NULL; 7117 uint32_t size = sizeof(volume); 7118 volume[0] = *left; 7119 volume[1] = *right; 7120 if (controller) { 7121 pVolume = volume; 7122 } 7123 status = (*mEffectInterface)->command(mEffectInterface, 7124 EFFECT_CMD_SET_VOLUME, 7125 size, 7126 volume, 7127 &size, 7128 pVolume); 7129 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7130 *left = volume[0]; 7131 *right = volume[1]; 7132 } 7133 } 7134 return status; 7135} 7136 7137status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7138{ 7139 Mutex::Autolock _l(mLock); 7140 status_t status = NO_ERROR; 7141 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7142 // audio pre processing modules on RecordThread can receive both output and 7143 // input device indication in the same call 7144 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7145 if (dev) { 7146 status_t cmdStatus; 7147 uint32_t size = sizeof(status_t); 7148 7149 status = (*mEffectInterface)->command(mEffectInterface, 7150 EFFECT_CMD_SET_DEVICE, 7151 sizeof(uint32_t), 7152 &dev, 7153 &size, 7154 &cmdStatus); 7155 if (status == NO_ERROR) { 7156 status = cmdStatus; 7157 } 7158 } 7159 dev = device & AUDIO_DEVICE_IN_ALL; 7160 if (dev) { 7161 status_t cmdStatus; 7162 uint32_t size = sizeof(status_t); 7163 7164 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7165 EFFECT_CMD_SET_INPUT_DEVICE, 7166 sizeof(uint32_t), 7167 &dev, 7168 &size, 7169 &cmdStatus); 7170 if (status2 == NO_ERROR) { 7171 status2 = cmdStatus; 7172 } 7173 if (status == NO_ERROR) { 7174 status = status2; 7175 } 7176 } 7177 } 7178 return status; 7179} 7180 7181status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7182{ 7183 Mutex::Autolock _l(mLock); 7184 status_t status = NO_ERROR; 7185 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7186 status_t cmdStatus; 7187 uint32_t size = sizeof(status_t); 7188 status = (*mEffectInterface)->command(mEffectInterface, 7189 EFFECT_CMD_SET_AUDIO_MODE, 7190 sizeof(audio_mode_t), 7191 &mode, 7192 &size, 7193 &cmdStatus); 7194 if (status == NO_ERROR) { 7195 status = cmdStatus; 7196 } 7197 } 7198 return status; 7199} 7200 7201void AudioFlinger::EffectModule::setSuspended(bool suspended) 7202{ 7203 Mutex::Autolock _l(mLock); 7204 mSuspended = suspended; 7205} 7206 7207bool AudioFlinger::EffectModule::suspended() const 7208{ 7209 Mutex::Autolock _l(mLock); 7210 return mSuspended; 7211} 7212 7213status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7214{ 7215 const size_t SIZE = 256; 7216 char buffer[SIZE]; 7217 String8 result; 7218 7219 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7220 result.append(buffer); 7221 7222 bool locked = tryLock(mLock); 7223 // failed to lock - AudioFlinger is probably deadlocked 7224 if (!locked) { 7225 result.append("\t\tCould not lock Fx mutex:\n"); 7226 } 7227 7228 result.append("\t\tSession Status State Engine:\n"); 7229 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7230 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7231 result.append(buffer); 7232 7233 result.append("\t\tDescriptor:\n"); 7234 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7235 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7236 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7237 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7238 result.append(buffer); 7239 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7240 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7241 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7242 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7243 result.append(buffer); 7244 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7245 mDescriptor.apiVersion, 7246 mDescriptor.flags); 7247 result.append(buffer); 7248 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7249 mDescriptor.name); 7250 result.append(buffer); 7251 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7252 mDescriptor.implementor); 7253 result.append(buffer); 7254 7255 result.append("\t\t- Input configuration:\n"); 7256 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7257 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7258 (uint32_t)mConfig.inputCfg.buffer.raw, 7259 mConfig.inputCfg.buffer.frameCount, 7260 mConfig.inputCfg.samplingRate, 7261 mConfig.inputCfg.channels, 7262 mConfig.inputCfg.format); 7263 result.append(buffer); 7264 7265 result.append("\t\t- Output configuration:\n"); 7266 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7267 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7268 (uint32_t)mConfig.outputCfg.buffer.raw, 7269 mConfig.outputCfg.buffer.frameCount, 7270 mConfig.outputCfg.samplingRate, 7271 mConfig.outputCfg.channels, 7272 mConfig.outputCfg.format); 7273 result.append(buffer); 7274 7275 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7276 result.append(buffer); 7277 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7278 for (size_t i = 0; i < mHandles.size(); ++i) { 7279 sp<EffectHandle> handle = mHandles[i].promote(); 7280 if (handle != 0) { 7281 handle->dump(buffer, SIZE); 7282 result.append(buffer); 7283 } 7284 } 7285 7286 result.append("\n"); 7287 7288 write(fd, result.string(), result.length()); 7289 7290 if (locked) { 7291 mLock.unlock(); 7292 } 7293 7294 return NO_ERROR; 7295} 7296 7297// ---------------------------------------------------------------------------- 7298// EffectHandle implementation 7299// ---------------------------------------------------------------------------- 7300 7301#undef LOG_TAG 7302#define LOG_TAG "AudioFlinger::EffectHandle" 7303 7304AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7305 const sp<AudioFlinger::Client>& client, 7306 const sp<IEffectClient>& effectClient, 7307 int32_t priority) 7308 : BnEffect(), 7309 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7310 mPriority(priority), mHasControl(false), mEnabled(false) 7311{ 7312 ALOGV("constructor %p", this); 7313 7314 if (client == 0) { 7315 return; 7316 } 7317 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7318 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7319 if (mCblkMemory != 0) { 7320 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7321 7322 if (mCblk != NULL) { 7323 new(mCblk) effect_param_cblk_t(); 7324 mBuffer = (uint8_t *)mCblk + bufOffset; 7325 } 7326 } else { 7327 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7328 return; 7329 } 7330} 7331 7332AudioFlinger::EffectHandle::~EffectHandle() 7333{ 7334 ALOGV("Destructor %p", this); 7335 disconnect(false); 7336 ALOGV("Destructor DONE %p", this); 7337} 7338 7339status_t AudioFlinger::EffectHandle::enable() 7340{ 7341 ALOGV("enable %p", this); 7342 if (!mHasControl) return INVALID_OPERATION; 7343 if (mEffect == 0) return DEAD_OBJECT; 7344 7345 if (mEnabled) { 7346 return NO_ERROR; 7347 } 7348 7349 mEnabled = true; 7350 7351 sp<ThreadBase> thread = mEffect->thread().promote(); 7352 if (thread != 0) { 7353 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7354 } 7355 7356 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7357 if (mEffect->suspended()) { 7358 return NO_ERROR; 7359 } 7360 7361 status_t status = mEffect->setEnabled(true); 7362 if (status != NO_ERROR) { 7363 if (thread != 0) { 7364 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7365 } 7366 mEnabled = false; 7367 } 7368 return status; 7369} 7370 7371status_t AudioFlinger::EffectHandle::disable() 7372{ 7373 ALOGV("disable %p", this); 7374 if (!mHasControl) return INVALID_OPERATION; 7375 if (mEffect == 0) return DEAD_OBJECT; 7376 7377 if (!mEnabled) { 7378 return NO_ERROR; 7379 } 7380 mEnabled = false; 7381 7382 if (mEffect->suspended()) { 7383 return NO_ERROR; 7384 } 7385 7386 status_t status = mEffect->setEnabled(false); 7387 7388 sp<ThreadBase> thread = mEffect->thread().promote(); 7389 if (thread != 0) { 7390 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7391 } 7392 7393 return status; 7394} 7395 7396void AudioFlinger::EffectHandle::disconnect() 7397{ 7398 disconnect(true); 7399} 7400 7401void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7402{ 7403 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7404 if (mEffect == 0) { 7405 return; 7406 } 7407 mEffect->disconnect(this, unpinIfLast); 7408 7409 if (mHasControl && mEnabled) { 7410 sp<ThreadBase> thread = mEffect->thread().promote(); 7411 if (thread != 0) { 7412 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7413 } 7414 } 7415 7416 // release sp on module => module destructor can be called now 7417 mEffect.clear(); 7418 if (mClient != 0) { 7419 if (mCblk != NULL) { 7420 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7421 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7422 } 7423 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7424 // Client destructor must run with AudioFlinger mutex locked 7425 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7426 mClient.clear(); 7427 } 7428} 7429 7430status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7431 uint32_t cmdSize, 7432 void *pCmdData, 7433 uint32_t *replySize, 7434 void *pReplyData) 7435{ 7436// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7437// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7438 7439 // only get parameter command is permitted for applications not controlling the effect 7440 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7441 return INVALID_OPERATION; 7442 } 7443 if (mEffect == 0) return DEAD_OBJECT; 7444 if (mClient == 0) return INVALID_OPERATION; 7445 7446 // handle commands that are not forwarded transparently to effect engine 7447 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7448 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7449 // no risk to block the whole media server process or mixer threads is we are stuck here 7450 Mutex::Autolock _l(mCblk->lock); 7451 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7452 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7453 mCblk->serverIndex = 0; 7454 mCblk->clientIndex = 0; 7455 return BAD_VALUE; 7456 } 7457 status_t status = NO_ERROR; 7458 while (mCblk->serverIndex < mCblk->clientIndex) { 7459 int reply; 7460 uint32_t rsize = sizeof(int); 7461 int *p = (int *)(mBuffer + mCblk->serverIndex); 7462 int size = *p++; 7463 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7464 ALOGW("command(): invalid parameter block size"); 7465 break; 7466 } 7467 effect_param_t *param = (effect_param_t *)p; 7468 if (param->psize == 0 || param->vsize == 0) { 7469 ALOGW("command(): null parameter or value size"); 7470 mCblk->serverIndex += size; 7471 continue; 7472 } 7473 uint32_t psize = sizeof(effect_param_t) + 7474 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7475 param->vsize; 7476 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7477 psize, 7478 p, 7479 &rsize, 7480 &reply); 7481 // stop at first error encountered 7482 if (ret != NO_ERROR) { 7483 status = ret; 7484 *(int *)pReplyData = reply; 7485 break; 7486 } else if (reply != NO_ERROR) { 7487 *(int *)pReplyData = reply; 7488 break; 7489 } 7490 mCblk->serverIndex += size; 7491 } 7492 mCblk->serverIndex = 0; 7493 mCblk->clientIndex = 0; 7494 return status; 7495 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7496 *(int *)pReplyData = NO_ERROR; 7497 return enable(); 7498 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7499 *(int *)pReplyData = NO_ERROR; 7500 return disable(); 7501 } 7502 7503 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7504} 7505 7506void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7507{ 7508 ALOGV("setControl %p control %d", this, hasControl); 7509 7510 mHasControl = hasControl; 7511 mEnabled = enabled; 7512 7513 if (signal && mEffectClient != 0) { 7514 mEffectClient->controlStatusChanged(hasControl); 7515 } 7516} 7517 7518void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7519 uint32_t cmdSize, 7520 void *pCmdData, 7521 uint32_t replySize, 7522 void *pReplyData) 7523{ 7524 if (mEffectClient != 0) { 7525 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7526 } 7527} 7528 7529 7530 7531void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7532{ 7533 if (mEffectClient != 0) { 7534 mEffectClient->enableStatusChanged(enabled); 7535 } 7536} 7537 7538status_t AudioFlinger::EffectHandle::onTransact( 7539 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7540{ 7541 return BnEffect::onTransact(code, data, reply, flags); 7542} 7543 7544 7545void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7546{ 7547 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7548 7549 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7550 (mClient == 0) ? getpid_cached : mClient->pid(), 7551 mPriority, 7552 mHasControl, 7553 !locked, 7554 mCblk ? mCblk->clientIndex : 0, 7555 mCblk ? mCblk->serverIndex : 0 7556 ); 7557 7558 if (locked) { 7559 mCblk->lock.unlock(); 7560 } 7561} 7562 7563#undef LOG_TAG 7564#define LOG_TAG "AudioFlinger::EffectChain" 7565 7566AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7567 int sessionId) 7568 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7569 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7570 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7571{ 7572 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7573 if (thread == NULL) { 7574 return; 7575 } 7576 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7577 thread->frameCount(); 7578} 7579 7580AudioFlinger::EffectChain::~EffectChain() 7581{ 7582 if (mOwnInBuffer) { 7583 delete mInBuffer; 7584 } 7585 7586} 7587 7588// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7589sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7590{ 7591 size_t size = mEffects.size(); 7592 7593 for (size_t i = 0; i < size; i++) { 7594 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7595 return mEffects[i]; 7596 } 7597 } 7598 return 0; 7599} 7600 7601// getEffectFromId_l() must be called with ThreadBase::mLock held 7602sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7603{ 7604 size_t size = mEffects.size(); 7605 7606 for (size_t i = 0; i < size; i++) { 7607 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7608 if (id == 0 || mEffects[i]->id() == id) { 7609 return mEffects[i]; 7610 } 7611 } 7612 return 0; 7613} 7614 7615// getEffectFromType_l() must be called with ThreadBase::mLock held 7616sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7617 const effect_uuid_t *type) 7618{ 7619 size_t size = mEffects.size(); 7620 7621 for (size_t i = 0; i < size; i++) { 7622 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7623 return mEffects[i]; 7624 } 7625 } 7626 return 0; 7627} 7628 7629// Must be called with EffectChain::mLock locked 7630void AudioFlinger::EffectChain::process_l() 7631{ 7632 sp<ThreadBase> thread = mThread.promote(); 7633 if (thread == 0) { 7634 ALOGW("process_l(): cannot promote mixer thread"); 7635 return; 7636 } 7637 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7638 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7639 // always process effects unless no more tracks are on the session and the effect tail 7640 // has been rendered 7641 bool doProcess = true; 7642 if (!isGlobalSession) { 7643 bool tracksOnSession = (trackCnt() != 0); 7644 7645 if (!tracksOnSession && mTailBufferCount == 0) { 7646 doProcess = false; 7647 } 7648 7649 if (activeTrackCnt() == 0) { 7650 // if no track is active and the effect tail has not been rendered, 7651 // the input buffer must be cleared here as the mixer process will not do it 7652 if (tracksOnSession || mTailBufferCount > 0) { 7653 size_t numSamples = thread->frameCount() * thread->channelCount(); 7654 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7655 if (mTailBufferCount > 0) { 7656 mTailBufferCount--; 7657 } 7658 } 7659 } 7660 } 7661 7662 size_t size = mEffects.size(); 7663 if (doProcess) { 7664 for (size_t i = 0; i < size; i++) { 7665 mEffects[i]->process(); 7666 } 7667 } 7668 for (size_t i = 0; i < size; i++) { 7669 mEffects[i]->updateState(); 7670 } 7671} 7672 7673// addEffect_l() must be called with PlaybackThread::mLock held 7674status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7675{ 7676 effect_descriptor_t desc = effect->desc(); 7677 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7678 7679 Mutex::Autolock _l(mLock); 7680 effect->setChain(this); 7681 sp<ThreadBase> thread = mThread.promote(); 7682 if (thread == 0) { 7683 return NO_INIT; 7684 } 7685 effect->setThread(thread); 7686 7687 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7688 // Auxiliary effects are inserted at the beginning of mEffects vector as 7689 // they are processed first and accumulated in chain input buffer 7690 mEffects.insertAt(effect, 0); 7691 7692 // the input buffer for auxiliary effect contains mono samples in 7693 // 32 bit format. This is to avoid saturation in AudoMixer 7694 // accumulation stage. Saturation is done in EffectModule::process() before 7695 // calling the process in effect engine 7696 size_t numSamples = thread->frameCount(); 7697 int32_t *buffer = new int32_t[numSamples]; 7698 memset(buffer, 0, numSamples * sizeof(int32_t)); 7699 effect->setInBuffer((int16_t *)buffer); 7700 // auxiliary effects output samples to chain input buffer for further processing 7701 // by insert effects 7702 effect->setOutBuffer(mInBuffer); 7703 } else { 7704 // Insert effects are inserted at the end of mEffects vector as they are processed 7705 // after track and auxiliary effects. 7706 // Insert effect order as a function of indicated preference: 7707 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7708 // another effect is present 7709 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7710 // last effect claiming first position 7711 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7712 // first effect claiming last position 7713 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7714 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7715 // already present 7716 7717 size_t size = mEffects.size(); 7718 size_t idx_insert = size; 7719 ssize_t idx_insert_first = -1; 7720 ssize_t idx_insert_last = -1; 7721 7722 for (size_t i = 0; i < size; i++) { 7723 effect_descriptor_t d = mEffects[i]->desc(); 7724 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7725 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7726 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7727 // check invalid effect chaining combinations 7728 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7729 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7730 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7731 return INVALID_OPERATION; 7732 } 7733 // remember position of first insert effect and by default 7734 // select this as insert position for new effect 7735 if (idx_insert == size) { 7736 idx_insert = i; 7737 } 7738 // remember position of last insert effect claiming 7739 // first position 7740 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7741 idx_insert_first = i; 7742 } 7743 // remember position of first insert effect claiming 7744 // last position 7745 if (iPref == EFFECT_FLAG_INSERT_LAST && 7746 idx_insert_last == -1) { 7747 idx_insert_last = i; 7748 } 7749 } 7750 } 7751 7752 // modify idx_insert from first position if needed 7753 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7754 if (idx_insert_last != -1) { 7755 idx_insert = idx_insert_last; 7756 } else { 7757 idx_insert = size; 7758 } 7759 } else { 7760 if (idx_insert_first != -1) { 7761 idx_insert = idx_insert_first + 1; 7762 } 7763 } 7764 7765 // always read samples from chain input buffer 7766 effect->setInBuffer(mInBuffer); 7767 7768 // if last effect in the chain, output samples to chain 7769 // output buffer, otherwise to chain input buffer 7770 if (idx_insert == size) { 7771 if (idx_insert != 0) { 7772 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7773 mEffects[idx_insert-1]->configure(); 7774 } 7775 effect->setOutBuffer(mOutBuffer); 7776 } else { 7777 effect->setOutBuffer(mInBuffer); 7778 } 7779 mEffects.insertAt(effect, idx_insert); 7780 7781 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7782 } 7783 effect->configure(); 7784 return NO_ERROR; 7785} 7786 7787// removeEffect_l() must be called with PlaybackThread::mLock held 7788size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7789{ 7790 Mutex::Autolock _l(mLock); 7791 size_t size = mEffects.size(); 7792 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7793 7794 for (size_t i = 0; i < size; i++) { 7795 if (effect == mEffects[i]) { 7796 // calling stop here will remove pre-processing effect from the audio HAL. 7797 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7798 // the middle of a read from audio HAL 7799 if (mEffects[i]->state() == EffectModule::ACTIVE || 7800 mEffects[i]->state() == EffectModule::STOPPING) { 7801 mEffects[i]->stop(); 7802 } 7803 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7804 delete[] effect->inBuffer(); 7805 } else { 7806 if (i == size - 1 && i != 0) { 7807 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7808 mEffects[i - 1]->configure(); 7809 } 7810 } 7811 mEffects.removeAt(i); 7812 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7813 break; 7814 } 7815 } 7816 7817 return mEffects.size(); 7818} 7819 7820// setDevice_l() must be called with PlaybackThread::mLock held 7821void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7822{ 7823 size_t size = mEffects.size(); 7824 for (size_t i = 0; i < size; i++) { 7825 mEffects[i]->setDevice(device); 7826 } 7827} 7828 7829// setMode_l() must be called with PlaybackThread::mLock held 7830void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7831{ 7832 size_t size = mEffects.size(); 7833 for (size_t i = 0; i < size; i++) { 7834 mEffects[i]->setMode(mode); 7835 } 7836} 7837 7838// setVolume_l() must be called with PlaybackThread::mLock held 7839bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7840{ 7841 uint32_t newLeft = *left; 7842 uint32_t newRight = *right; 7843 bool hasControl = false; 7844 int ctrlIdx = -1; 7845 size_t size = mEffects.size(); 7846 7847 // first update volume controller 7848 for (size_t i = size; i > 0; i--) { 7849 if (mEffects[i - 1]->isProcessEnabled() && 7850 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7851 ctrlIdx = i - 1; 7852 hasControl = true; 7853 break; 7854 } 7855 } 7856 7857 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7858 if (hasControl) { 7859 *left = mNewLeftVolume; 7860 *right = mNewRightVolume; 7861 } 7862 return hasControl; 7863 } 7864 7865 mVolumeCtrlIdx = ctrlIdx; 7866 mLeftVolume = newLeft; 7867 mRightVolume = newRight; 7868 7869 // second get volume update from volume controller 7870 if (ctrlIdx >= 0) { 7871 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7872 mNewLeftVolume = newLeft; 7873 mNewRightVolume = newRight; 7874 } 7875 // then indicate volume to all other effects in chain. 7876 // Pass altered volume to effects before volume controller 7877 // and requested volume to effects after controller 7878 uint32_t lVol = newLeft; 7879 uint32_t rVol = newRight; 7880 7881 for (size_t i = 0; i < size; i++) { 7882 if ((int)i == ctrlIdx) continue; 7883 // this also works for ctrlIdx == -1 when there is no volume controller 7884 if ((int)i > ctrlIdx) { 7885 lVol = *left; 7886 rVol = *right; 7887 } 7888 mEffects[i]->setVolume(&lVol, &rVol, false); 7889 } 7890 *left = newLeft; 7891 *right = newRight; 7892 7893 return hasControl; 7894} 7895 7896status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7897{ 7898 const size_t SIZE = 256; 7899 char buffer[SIZE]; 7900 String8 result; 7901 7902 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7903 result.append(buffer); 7904 7905 bool locked = tryLock(mLock); 7906 // failed to lock - AudioFlinger is probably deadlocked 7907 if (!locked) { 7908 result.append("\tCould not lock mutex:\n"); 7909 } 7910 7911 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7912 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7913 mEffects.size(), 7914 (uint32_t)mInBuffer, 7915 (uint32_t)mOutBuffer, 7916 mActiveTrackCnt); 7917 result.append(buffer); 7918 write(fd, result.string(), result.size()); 7919 7920 for (size_t i = 0; i < mEffects.size(); ++i) { 7921 sp<EffectModule> effect = mEffects[i]; 7922 if (effect != 0) { 7923 effect->dump(fd, args); 7924 } 7925 } 7926 7927 if (locked) { 7928 mLock.unlock(); 7929 } 7930 7931 return NO_ERROR; 7932} 7933 7934// must be called with ThreadBase::mLock held 7935void AudioFlinger::EffectChain::setEffectSuspended_l( 7936 const effect_uuid_t *type, bool suspend) 7937{ 7938 sp<SuspendedEffectDesc> desc; 7939 // use effect type UUID timelow as key as there is no real risk of identical 7940 // timeLow fields among effect type UUIDs. 7941 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7942 if (suspend) { 7943 if (index >= 0) { 7944 desc = mSuspendedEffects.valueAt(index); 7945 } else { 7946 desc = new SuspendedEffectDesc(); 7947 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7948 mSuspendedEffects.add(type->timeLow, desc); 7949 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7950 } 7951 if (desc->mRefCount++ == 0) { 7952 sp<EffectModule> effect = getEffectIfEnabled(type); 7953 if (effect != 0) { 7954 desc->mEffect = effect; 7955 effect->setSuspended(true); 7956 effect->setEnabled(false); 7957 } 7958 } 7959 } else { 7960 if (index < 0) { 7961 return; 7962 } 7963 desc = mSuspendedEffects.valueAt(index); 7964 if (desc->mRefCount <= 0) { 7965 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7966 desc->mRefCount = 1; 7967 } 7968 if (--desc->mRefCount == 0) { 7969 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7970 if (desc->mEffect != 0) { 7971 sp<EffectModule> effect = desc->mEffect.promote(); 7972 if (effect != 0) { 7973 effect->setSuspended(false); 7974 sp<EffectHandle> handle = effect->controlHandle(); 7975 if (handle != 0) { 7976 effect->setEnabled(handle->enabled()); 7977 } 7978 } 7979 desc->mEffect.clear(); 7980 } 7981 mSuspendedEffects.removeItemsAt(index); 7982 } 7983 } 7984} 7985 7986// must be called with ThreadBase::mLock held 7987void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7988{ 7989 sp<SuspendedEffectDesc> desc; 7990 7991 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7992 if (suspend) { 7993 if (index >= 0) { 7994 desc = mSuspendedEffects.valueAt(index); 7995 } else { 7996 desc = new SuspendedEffectDesc(); 7997 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7998 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7999 } 8000 if (desc->mRefCount++ == 0) { 8001 Vector< sp<EffectModule> > effects; 8002 getSuspendEligibleEffects(effects); 8003 for (size_t i = 0; i < effects.size(); i++) { 8004 setEffectSuspended_l(&effects[i]->desc().type, true); 8005 } 8006 } 8007 } else { 8008 if (index < 0) { 8009 return; 8010 } 8011 desc = mSuspendedEffects.valueAt(index); 8012 if (desc->mRefCount <= 0) { 8013 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8014 desc->mRefCount = 1; 8015 } 8016 if (--desc->mRefCount == 0) { 8017 Vector<const effect_uuid_t *> types; 8018 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8019 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8020 continue; 8021 } 8022 types.add(&mSuspendedEffects.valueAt(i)->mType); 8023 } 8024 for (size_t i = 0; i < types.size(); i++) { 8025 setEffectSuspended_l(types[i], false); 8026 } 8027 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8028 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8029 } 8030 } 8031} 8032 8033 8034// The volume effect is used for automated tests only 8035#ifndef OPENSL_ES_H_ 8036static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8037 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8038const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8039#endif //OPENSL_ES_H_ 8040 8041bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8042{ 8043 // auxiliary effects and visualizer are never suspended on output mix 8044 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8045 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8046 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8047 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8048 return false; 8049 } 8050 return true; 8051} 8052 8053void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8054{ 8055 effects.clear(); 8056 for (size_t i = 0; i < mEffects.size(); i++) { 8057 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8058 effects.add(mEffects[i]); 8059 } 8060 } 8061} 8062 8063sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8064 const effect_uuid_t *type) 8065{ 8066 sp<EffectModule> effect = getEffectFromType_l(type); 8067 return effect != 0 && effect->isEnabled() ? effect : 0; 8068} 8069 8070void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8071 bool enabled) 8072{ 8073 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8074 if (enabled) { 8075 if (index < 0) { 8076 // if the effect is not suspend check if all effects are suspended 8077 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8078 if (index < 0) { 8079 return; 8080 } 8081 if (!isEffectEligibleForSuspend(effect->desc())) { 8082 return; 8083 } 8084 setEffectSuspended_l(&effect->desc().type, enabled); 8085 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8086 if (index < 0) { 8087 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8088 return; 8089 } 8090 } 8091 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8092 effect->desc().type.timeLow); 8093 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8094 // if effect is requested to suspended but was not yet enabled, supend it now. 8095 if (desc->mEffect == 0) { 8096 desc->mEffect = effect; 8097 effect->setEnabled(false); 8098 effect->setSuspended(true); 8099 } 8100 } else { 8101 if (index < 0) { 8102 return; 8103 } 8104 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8105 effect->desc().type.timeLow); 8106 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8107 desc->mEffect.clear(); 8108 effect->setSuspended(false); 8109 } 8110} 8111 8112#undef LOG_TAG 8113#define LOG_TAG "AudioFlinger" 8114 8115// ---------------------------------------------------------------------------- 8116 8117status_t AudioFlinger::onTransact( 8118 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8119{ 8120 return BnAudioFlinger::onTransact(code, data, reply, flags); 8121} 8122 8123}; // namespace android 8124