AudioFlinger.cpp revision a1472d9883e35edd280201c8be3191695007dfd4
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#undef ADD_BATTERY_DATA 41 42#ifdef ADD_BATTERY_DATA 43#include <media/IMediaPlayerService.h> 44#include <media/IMediaDeathNotifier.h> 45#endif 46 47#include <private/media/AudioTrackShared.h> 48#include <private/media/AudioEffectShared.h> 49 50#include <system/audio.h> 51#include <hardware/audio.h> 52 53#include "AudioMixer.h" 54#include "AudioFlinger.h" 55#include "ServiceUtilities.h" 56 57#include <media/EffectsFactoryApi.h> 58#include <audio_effects/effect_visualizer.h> 59#include <audio_effects/effect_ns.h> 60#include <audio_effects/effect_aec.h> 61 62#include <audio_utils/primitives.h> 63 64#include <powermanager/PowerManager.h> 65 66// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72#include <common_time/cc_helper.h> 73#include <common_time/local_clock.h> 74 75// ---------------------------------------------------------------------------- 76 77 78namespace android { 79 80static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 81static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 82 83static const float MAX_GAIN = 4096.0f; 84static const uint32_t MAX_GAIN_INT = 0x1000; 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95static const int kDumpLockRetries = 50; 96static const int kDumpLockSleepUs = 20000; 97 98// don't warn about blocked writes or record buffer overflows more often than this 99static const nsecs_t kWarningThrottleNs = seconds(5); 100 101// RecordThread loop sleep time upon application overrun or audio HAL read error 102static const int kRecordThreadSleepUs = 5000; 103 104// maximum time to wait for setParameters to complete 105static const nsecs_t kSetParametersTimeoutNs = seconds(2); 106 107// minimum sleep time for the mixer thread loop when tracks are active but in underrun 108static const uint32_t kMinThreadSleepTimeUs = 5000; 109// maximum divider applied to the active sleep time in the mixer thread loop 110static const uint32_t kMaxThreadSleepTimeShift = 2; 111 112nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 113 114// ---------------------------------------------------------------------------- 115 116#ifdef ADD_BATTERY_DATA 117// To collect the amplifier usage 118static void addBatteryData(uint32_t params) { 119 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 120 if (service == NULL) { 121 // it already logged 122 return; 123 } 124 125 service->addBatteryData(params); 126} 127#endif 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152static const char * const audio_interfaces[] = { 153 "primary", 154 "a2dp", 155 "usb", 156}; 157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 158 159// ---------------------------------------------------------------------------- 160 161AudioFlinger::AudioFlinger() 162 : BnAudioFlinger(), 163 mPrimaryHardwareDev(NULL), 164 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 165 mMasterVolume(1.0f), 166 mMasterVolumeSupportLvl(MVS_NONE), 167 mMasterMute(false), 168 mNextUniqueId(1), 169 mMode(AUDIO_MODE_INVALID), 170 mBtNrecIsOff(false) 171{ 172} 173 174void AudioFlinger::onFirstRef() 175{ 176 int rc = 0; 177 178 Mutex::Autolock _l(mLock); 179 180 /* TODO: move all this work into an Init() function */ 181 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 182 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 183 uint32_t int_val; 184 if (1 == sscanf(val_str, "%u", &int_val)) { 185 mStandbyTimeInNsecs = milliseconds(int_val); 186 ALOGI("Using %u mSec as standby time.", int_val); 187 } else { 188 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 189 ALOGI("Using default %u mSec as standby time.", 190 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 191 } 192 } 193 194 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 195 const hw_module_t *mod; 196 audio_hw_device_t *dev; 197 198 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 199 if (rc) 200 continue; 201 202 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 203 mod->name, mod->id); 204 mAudioHwDevs.push(dev); 205 206 if (mPrimaryHardwareDev == NULL) { 207 mPrimaryHardwareDev = dev; 208 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 209 mod->name, mod->id, audio_interfaces[i]); 210 } 211 } 212 213 if (mPrimaryHardwareDev == NULL) { 214 ALOGE("Primary audio interface not found"); 215 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 216 } 217 218 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 219 // primary HW dev is selected can change so these conditions might not always be equivalent. 220 // When that happens, re-visit all the code that assumes this. 221 222 AutoMutex lock(mHardwareLock); 223 224 // Determine the level of master volume support the primary audio HAL has, 225 // and set the initial master volume at the same time. 226 float initialVolume = 1.0; 227 mMasterVolumeSupportLvl = MVS_NONE; 228 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 229 audio_hw_device_t *dev = mPrimaryHardwareDev; 230 231 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 232 if ((NULL != dev->get_master_volume) && 233 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 234 mMasterVolumeSupportLvl = MVS_FULL; 235 } else { 236 mMasterVolumeSupportLvl = MVS_SETONLY; 237 initialVolume = 1.0; 238 } 239 240 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 241 if ((NULL == dev->set_master_volume) || 242 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 243 mMasterVolumeSupportLvl = MVS_NONE; 244 } 245 mHardwareStatus = AUDIO_HW_IDLE; 246 } 247 248 // Set the mode for each audio HAL, and try to set the initial volume (if 249 // supported) for all of the non-primary audio HALs. 250 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 251 audio_hw_device_t *dev = mAudioHwDevs[i]; 252 253 mHardwareStatus = AUDIO_HW_INIT; 254 rc = dev->init_check(dev); 255 mHardwareStatus = AUDIO_HW_IDLE; 256 if (rc == 0) { 257 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 258 mHardwareStatus = AUDIO_HW_SET_MODE; 259 dev->set_mode(dev, mMode); 260 261 if ((dev != mPrimaryHardwareDev) && 262 (NULL != dev->set_master_volume)) { 263 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 264 dev->set_master_volume(dev, initialVolume); 265 } 266 267 mHardwareStatus = AUDIO_HW_IDLE; 268 } 269 } 270 271 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 272 ? initialVolume 273 : 1.0; 274 mMasterVolume = initialVolume; 275 mHardwareStatus = AUDIO_HW_IDLE; 276} 277 278AudioFlinger::~AudioFlinger() 279{ 280 281 while (!mRecordThreads.isEmpty()) { 282 // closeInput() will remove first entry from mRecordThreads 283 closeInput(mRecordThreads.keyAt(0)); 284 } 285 while (!mPlaybackThreads.isEmpty()) { 286 // closeOutput() will remove first entry from mPlaybackThreads 287 closeOutput(mPlaybackThreads.keyAt(0)); 288 } 289 290 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 291 // no mHardwareLock needed, as there are no other references to this 292 audio_hw_device_close(mAudioHwDevs[i]); 293 } 294} 295 296audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 297{ 298 /* first matching HW device is returned */ 299 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 300 audio_hw_device_t *dev = mAudioHwDevs[i]; 301 if ((dev->get_supported_devices(dev) & devices) == devices) 302 return dev; 303 } 304 return NULL; 305} 306 307status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 308{ 309 const size_t SIZE = 256; 310 char buffer[SIZE]; 311 String8 result; 312 313 result.append("Clients:\n"); 314 for (size_t i = 0; i < mClients.size(); ++i) { 315 sp<Client> client = mClients.valueAt(i).promote(); 316 if (client != 0) { 317 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 318 result.append(buffer); 319 } 320 } 321 322 result.append("Global session refs:\n"); 323 result.append(" session pid count\n"); 324 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 325 AudioSessionRef *r = mAudioSessionRefs[i]; 326 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 327 result.append(buffer); 328 } 329 write(fd, result.string(), result.size()); 330 return NO_ERROR; 331} 332 333 334status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 335{ 336 const size_t SIZE = 256; 337 char buffer[SIZE]; 338 String8 result; 339 hardware_call_state hardwareStatus = mHardwareStatus; 340 341 snprintf(buffer, SIZE, "Hardware status: %d\n" 342 "Standby Time mSec: %u\n", 343 hardwareStatus, 344 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 345 result.append(buffer); 346 write(fd, result.string(), result.size()); 347 return NO_ERROR; 348} 349 350status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 351{ 352 const size_t SIZE = 256; 353 char buffer[SIZE]; 354 String8 result; 355 snprintf(buffer, SIZE, "Permission Denial: " 356 "can't dump AudioFlinger from pid=%d, uid=%d\n", 357 IPCThreadState::self()->getCallingPid(), 358 IPCThreadState::self()->getCallingUid()); 359 result.append(buffer); 360 write(fd, result.string(), result.size()); 361 return NO_ERROR; 362} 363 364static bool tryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = tryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = tryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 dumpClients(fd, args); 400 dumpInternals(fd, args); 401 402 // dump playback threads 403 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 404 mPlaybackThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump record threads 408 for (size_t i = 0; i < mRecordThreads.size(); i++) { 409 mRecordThreads.valueAt(i)->dump(fd, args); 410 } 411 412 // dump all hardware devs 413 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 414 audio_hw_device_t *dev = mAudioHwDevs[i]; 415 dev->dump(dev, fd); 416 } 417 if (locked) mLock.unlock(); 418 } 419 return NO_ERROR; 420} 421 422sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 423{ 424 // If pid is already in the mClients wp<> map, then use that entry 425 // (for which promote() is always != 0), otherwise create a new entry and Client. 426 sp<Client> client = mClients.valueFor(pid).promote(); 427 if (client == 0) { 428 client = new Client(this, pid); 429 mClients.add(pid, client); 430 } 431 432 return client; 433} 434 435// IAudioFlinger interface 436 437 438sp<IAudioTrack> AudioFlinger::createTrack( 439 pid_t pid, 440 audio_stream_type_t streamType, 441 uint32_t sampleRate, 442 audio_format_t format, 443 uint32_t channelMask, 444 int frameCount, 445 IAudioFlinger::track_flags_t flags, 446 const sp<IMemory>& sharedBuffer, 447 audio_io_handle_t output, 448 int *sessionId, 449 status_t *status) 450{ 451 sp<PlaybackThread::Track> track; 452 sp<TrackHandle> trackHandle; 453 sp<Client> client; 454 status_t lStatus; 455 int lSessionId; 456 457 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 458 // but if someone uses binder directly they could bypass that and cause us to crash 459 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 460 ALOGE("createTrack() invalid stream type %d", streamType); 461 lStatus = BAD_VALUE; 462 goto Exit; 463 } 464 465 { 466 Mutex::Autolock _l(mLock); 467 PlaybackThread *thread = checkPlaybackThread_l(output); 468 PlaybackThread *effectThread = NULL; 469 if (thread == NULL) { 470 ALOGE("unknown output thread"); 471 lStatus = BAD_VALUE; 472 goto Exit; 473 } 474 475 client = registerPid_l(pid); 476 477 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 478 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 479 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 480 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 481 if (mPlaybackThreads.keyAt(i) != output) { 482 // prevent same audio session on different output threads 483 uint32_t sessions = t->hasAudioSession(*sessionId); 484 if (sessions & PlaybackThread::TRACK_SESSION) { 485 ALOGE("createTrack() session ID %d already in use", *sessionId); 486 lStatus = BAD_VALUE; 487 goto Exit; 488 } 489 // check if an effect with same session ID is waiting for a track to be created 490 if (sessions & PlaybackThread::EFFECT_SESSION) { 491 effectThread = t.get(); 492 } 493 } 494 } 495 lSessionId = *sessionId; 496 } else { 497 // if no audio session id is provided, create one here 498 lSessionId = nextUniqueId(); 499 if (sessionId != NULL) { 500 *sessionId = lSessionId; 501 } 502 } 503 ALOGV("createTrack() lSessionId: %d", lSessionId); 504 505 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 506 track = thread->createTrack_l(client, streamType, sampleRate, format, 507 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 508 509 // move effect chain to this output thread if an effect on same session was waiting 510 // for a track to be created 511 if (lStatus == NO_ERROR && effectThread != NULL) { 512 Mutex::Autolock _dl(thread->mLock); 513 Mutex::Autolock _sl(effectThread->mLock); 514 moveEffectChain_l(lSessionId, effectThread, thread, true); 515 } 516 517 // Look for sync events awaiting for a session to be used. 518 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 519 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 520 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 521 track->setSyncEvent(mPendingSyncEvents[i]); 522 mPendingSyncEvents.removeAt(i); 523 i--; 524 } 525 } 526 } 527 } 528 if (lStatus == NO_ERROR) { 529 trackHandle = new TrackHandle(track); 530 } else { 531 // remove local strong reference to Client before deleting the Track so that the Client 532 // destructor is called by the TrackBase destructor with mLock held 533 client.clear(); 534 track.clear(); 535 } 536 537Exit: 538 if (status != NULL) { 539 *status = lStatus; 540 } 541 return trackHandle; 542} 543 544uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 545{ 546 Mutex::Autolock _l(mLock); 547 PlaybackThread *thread = checkPlaybackThread_l(output); 548 if (thread == NULL) { 549 ALOGW("sampleRate() unknown thread %d", output); 550 return 0; 551 } 552 return thread->sampleRate(); 553} 554 555int AudioFlinger::channelCount(audio_io_handle_t output) const 556{ 557 Mutex::Autolock _l(mLock); 558 PlaybackThread *thread = checkPlaybackThread_l(output); 559 if (thread == NULL) { 560 ALOGW("channelCount() unknown thread %d", output); 561 return 0; 562 } 563 return thread->channelCount(); 564} 565 566audio_format_t AudioFlinger::format(audio_io_handle_t output) const 567{ 568 Mutex::Autolock _l(mLock); 569 PlaybackThread *thread = checkPlaybackThread_l(output); 570 if (thread == NULL) { 571 ALOGW("format() unknown thread %d", output); 572 return AUDIO_FORMAT_INVALID; 573 } 574 return thread->format(); 575} 576 577size_t AudioFlinger::frameCount(audio_io_handle_t output) const 578{ 579 Mutex::Autolock _l(mLock); 580 PlaybackThread *thread = checkPlaybackThread_l(output); 581 if (thread == NULL) { 582 ALOGW("frameCount() unknown thread %d", output); 583 return 0; 584 } 585 return thread->frameCount(); 586} 587 588uint32_t AudioFlinger::latency(audio_io_handle_t output) const 589{ 590 Mutex::Autolock _l(mLock); 591 PlaybackThread *thread = checkPlaybackThread_l(output); 592 if (thread == NULL) { 593 ALOGW("latency() unknown thread %d", output); 594 return 0; 595 } 596 return thread->latency(); 597} 598 599status_t AudioFlinger::setMasterVolume(float value) 600{ 601 status_t ret = initCheck(); 602 if (ret != NO_ERROR) { 603 return ret; 604 } 605 606 // check calling permissions 607 if (!settingsAllowed()) { 608 return PERMISSION_DENIED; 609 } 610 611 float swmv = value; 612 613 // when hw supports master volume, don't scale in sw mixer 614 if (MVS_NONE != mMasterVolumeSupportLvl) { 615 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 616 AutoMutex lock(mHardwareLock); 617 audio_hw_device_t *dev = mAudioHwDevs[i]; 618 619 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 620 if (NULL != dev->set_master_volume) { 621 dev->set_master_volume(dev, value); 622 } 623 mHardwareStatus = AUDIO_HW_IDLE; 624 } 625 626 swmv = 1.0; 627 } 628 629 Mutex::Autolock _l(mLock); 630 mMasterVolume = value; 631 mMasterVolumeSW = swmv; 632 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 633 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 634 635 return NO_ERROR; 636} 637 638status_t AudioFlinger::setMode(audio_mode_t mode) 639{ 640 status_t ret = initCheck(); 641 if (ret != NO_ERROR) { 642 return ret; 643 } 644 645 // check calling permissions 646 if (!settingsAllowed()) { 647 return PERMISSION_DENIED; 648 } 649 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 650 ALOGW("Illegal value: setMode(%d)", mode); 651 return BAD_VALUE; 652 } 653 654 { // scope for the lock 655 AutoMutex lock(mHardwareLock); 656 mHardwareStatus = AUDIO_HW_SET_MODE; 657 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 658 mHardwareStatus = AUDIO_HW_IDLE; 659 } 660 661 if (NO_ERROR == ret) { 662 Mutex::Autolock _l(mLock); 663 mMode = mode; 664 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 665 mPlaybackThreads.valueAt(i)->setMode(mode); 666 } 667 668 return ret; 669} 670 671status_t AudioFlinger::setMicMute(bool state) 672{ 673 status_t ret = initCheck(); 674 if (ret != NO_ERROR) { 675 return ret; 676 } 677 678 // check calling permissions 679 if (!settingsAllowed()) { 680 return PERMISSION_DENIED; 681 } 682 683 AutoMutex lock(mHardwareLock); 684 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 685 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 686 mHardwareStatus = AUDIO_HW_IDLE; 687 return ret; 688} 689 690bool AudioFlinger::getMicMute() const 691{ 692 status_t ret = initCheck(); 693 if (ret != NO_ERROR) { 694 return false; 695 } 696 697 bool state = AUDIO_MODE_INVALID; 698 AutoMutex lock(mHardwareLock); 699 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 700 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 701 mHardwareStatus = AUDIO_HW_IDLE; 702 return state; 703} 704 705status_t AudioFlinger::setMasterMute(bool muted) 706{ 707 // check calling permissions 708 if (!settingsAllowed()) { 709 return PERMISSION_DENIED; 710 } 711 712 Mutex::Autolock _l(mLock); 713 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 714 mMasterMute = muted; 715 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 716 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 717 718 return NO_ERROR; 719} 720 721float AudioFlinger::masterVolume() const 722{ 723 Mutex::Autolock _l(mLock); 724 return masterVolume_l(); 725} 726 727float AudioFlinger::masterVolumeSW() const 728{ 729 Mutex::Autolock _l(mLock); 730 return masterVolumeSW_l(); 731} 732 733bool AudioFlinger::masterMute() const 734{ 735 Mutex::Autolock _l(mLock); 736 return masterMute_l(); 737} 738 739float AudioFlinger::masterVolume_l() const 740{ 741 if (MVS_FULL == mMasterVolumeSupportLvl) { 742 float ret_val; 743 AutoMutex lock(mHardwareLock); 744 745 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 746 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 747 (NULL != mPrimaryHardwareDev->get_master_volume), 748 "can't get master volume"); 749 750 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 751 mHardwareStatus = AUDIO_HW_IDLE; 752 return ret_val; 753 } 754 755 return mMasterVolume; 756} 757 758status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 759 audio_io_handle_t output) 760{ 761 // check calling permissions 762 if (!settingsAllowed()) { 763 return PERMISSION_DENIED; 764 } 765 766 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 767 ALOGE("setStreamVolume() invalid stream %d", stream); 768 return BAD_VALUE; 769 } 770 771 AutoMutex lock(mLock); 772 PlaybackThread *thread = NULL; 773 if (output) { 774 thread = checkPlaybackThread_l(output); 775 if (thread == NULL) { 776 return BAD_VALUE; 777 } 778 } 779 780 mStreamTypes[stream].volume = value; 781 782 if (thread == NULL) { 783 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 784 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 785 } 786 } else { 787 thread->setStreamVolume(stream, value); 788 } 789 790 return NO_ERROR; 791} 792 793status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 794{ 795 // check calling permissions 796 if (!settingsAllowed()) { 797 return PERMISSION_DENIED; 798 } 799 800 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 801 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 802 ALOGE("setStreamMute() invalid stream %d", stream); 803 return BAD_VALUE; 804 } 805 806 AutoMutex lock(mLock); 807 mStreamTypes[stream].mute = muted; 808 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 809 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 810 811 return NO_ERROR; 812} 813 814float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return 0.0f; 818 } 819 820 AutoMutex lock(mLock); 821 float volume; 822 if (output) { 823 PlaybackThread *thread = checkPlaybackThread_l(output); 824 if (thread == NULL) { 825 return 0.0f; 826 } 827 volume = thread->streamVolume(stream); 828 } else { 829 volume = streamVolume_l(stream); 830 } 831 832 return volume; 833} 834 835bool AudioFlinger::streamMute(audio_stream_type_t stream) const 836{ 837 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 838 return true; 839 } 840 841 AutoMutex lock(mLock); 842 return streamMute_l(stream); 843} 844 845status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 846{ 847 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 848 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 849 // check calling permissions 850 if (!settingsAllowed()) { 851 return PERMISSION_DENIED; 852 } 853 854 // ioHandle == 0 means the parameters are global to the audio hardware interface 855 if (ioHandle == 0) { 856 status_t final_result = NO_ERROR; 857 { 858 AutoMutex lock(mHardwareLock); 859 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 860 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 861 audio_hw_device_t *dev = mAudioHwDevs[i]; 862 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 863 final_result = result ?: final_result; 864 } 865 mHardwareStatus = AUDIO_HW_IDLE; 866 } 867 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 868 AudioParameter param = AudioParameter(keyValuePairs); 869 String8 value; 870 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 871 Mutex::Autolock _l(mLock); 872 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 873 if (mBtNrecIsOff != btNrecIsOff) { 874 for (size_t i = 0; i < mRecordThreads.size(); i++) { 875 sp<RecordThread> thread = mRecordThreads.valueAt(i); 876 RecordThread::RecordTrack *track = thread->track(); 877 if (track != NULL) { 878 audio_devices_t device = (audio_devices_t)( 879 thread->device() & AUDIO_DEVICE_IN_ALL); 880 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 881 thread->setEffectSuspended(FX_IID_AEC, 882 suspend, 883 track->sessionId()); 884 thread->setEffectSuspended(FX_IID_NS, 885 suspend, 886 track->sessionId()); 887 } 888 } 889 mBtNrecIsOff = btNrecIsOff; 890 } 891 } 892 return final_result; 893 } 894 895 // hold a strong ref on thread in case closeOutput() or closeInput() is called 896 // and the thread is exited once the lock is released 897 sp<ThreadBase> thread; 898 { 899 Mutex::Autolock _l(mLock); 900 thread = checkPlaybackThread_l(ioHandle); 901 if (thread == NULL) { 902 thread = checkRecordThread_l(ioHandle); 903 } else if (thread == primaryPlaybackThread_l()) { 904 // indicate output device change to all input threads for pre processing 905 AudioParameter param = AudioParameter(keyValuePairs); 906 int value; 907 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 908 (value != 0)) { 909 for (size_t i = 0; i < mRecordThreads.size(); i++) { 910 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 911 } 912 } 913 } 914 } 915 if (thread != 0) { 916 return thread->setParameters(keyValuePairs); 917 } 918 return BAD_VALUE; 919} 920 921String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 922{ 923// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 924// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 925 926 if (ioHandle == 0) { 927 String8 out_s8; 928 929 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 930 char *s; 931 { 932 AutoMutex lock(mHardwareLock); 933 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 934 audio_hw_device_t *dev = mAudioHwDevs[i]; 935 s = dev->get_parameters(dev, keys.string()); 936 mHardwareStatus = AUDIO_HW_IDLE; 937 } 938 out_s8 += String8(s ? s : ""); 939 free(s); 940 } 941 return out_s8; 942 } 943 944 Mutex::Autolock _l(mLock); 945 946 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 947 if (playbackThread != NULL) { 948 return playbackThread->getParameters(keys); 949 } 950 RecordThread *recordThread = checkRecordThread_l(ioHandle); 951 if (recordThread != NULL) { 952 return recordThread->getParameters(keys); 953 } 954 return String8(""); 955} 956 957size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 958{ 959 status_t ret = initCheck(); 960 if (ret != NO_ERROR) { 961 return 0; 962 } 963 964 AutoMutex lock(mHardwareLock); 965 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 966 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 967 mHardwareStatus = AUDIO_HW_IDLE; 968 return size; 969} 970 971unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 972{ 973 if (ioHandle == 0) { 974 return 0; 975 } 976 977 Mutex::Autolock _l(mLock); 978 979 RecordThread *recordThread = checkRecordThread_l(ioHandle); 980 if (recordThread != NULL) { 981 return recordThread->getInputFramesLost(); 982 } 983 return 0; 984} 985 986status_t AudioFlinger::setVoiceVolume(float value) 987{ 988 status_t ret = initCheck(); 989 if (ret != NO_ERROR) { 990 return ret; 991 } 992 993 // check calling permissions 994 if (!settingsAllowed()) { 995 return PERMISSION_DENIED; 996 } 997 998 AutoMutex lock(mHardwareLock); 999 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1000 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1001 mHardwareStatus = AUDIO_HW_IDLE; 1002 1003 return ret; 1004} 1005 1006status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1007 audio_io_handle_t output) const 1008{ 1009 status_t status; 1010 1011 Mutex::Autolock _l(mLock); 1012 1013 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1014 if (playbackThread != NULL) { 1015 return playbackThread->getRenderPosition(halFrames, dspFrames); 1016 } 1017 1018 return BAD_VALUE; 1019} 1020 1021void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1022{ 1023 1024 Mutex::Autolock _l(mLock); 1025 1026 pid_t pid = IPCThreadState::self()->getCallingPid(); 1027 if (mNotificationClients.indexOfKey(pid) < 0) { 1028 sp<NotificationClient> notificationClient = new NotificationClient(this, 1029 client, 1030 pid); 1031 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1032 1033 mNotificationClients.add(pid, notificationClient); 1034 1035 sp<IBinder> binder = client->asBinder(); 1036 binder->linkToDeath(notificationClient); 1037 1038 // the config change is always sent from playback or record threads to avoid deadlock 1039 // with AudioSystem::gLock 1040 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1041 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1042 } 1043 1044 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1045 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1046 } 1047 } 1048} 1049 1050void AudioFlinger::removeNotificationClient(pid_t pid) 1051{ 1052 Mutex::Autolock _l(mLock); 1053 1054 mNotificationClients.removeItem(pid); 1055 1056 ALOGV("%d died, releasing its sessions", pid); 1057 size_t num = mAudioSessionRefs.size(); 1058 bool removed = false; 1059 for (size_t i = 0; i< num; ) { 1060 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1061 ALOGV(" pid %d @ %d", ref->mPid, i); 1062 if (ref->mPid == pid) { 1063 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1064 mAudioSessionRefs.removeAt(i); 1065 delete ref; 1066 removed = true; 1067 num--; 1068 } else { 1069 i++; 1070 } 1071 } 1072 if (removed) { 1073 purgeStaleEffects_l(); 1074 } 1075} 1076 1077// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1078void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1079{ 1080 size_t size = mNotificationClients.size(); 1081 for (size_t i = 0; i < size; i++) { 1082 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1083 param2); 1084 } 1085} 1086 1087// removeClient_l() must be called with AudioFlinger::mLock held 1088void AudioFlinger::removeClient_l(pid_t pid) 1089{ 1090 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1091 mClients.removeItem(pid); 1092} 1093 1094 1095// ---------------------------------------------------------------------------- 1096 1097AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1098 uint32_t device, type_t type) 1099 : Thread(false), 1100 mType(type), 1101 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1102 // mChannelMask 1103 mChannelCount(0), 1104 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1105 mParamStatus(NO_ERROR), 1106 mStandby(false), mId(id), 1107 mDevice(device), 1108 mDeathRecipient(new PMDeathRecipient(this)) 1109{ 1110} 1111 1112AudioFlinger::ThreadBase::~ThreadBase() 1113{ 1114 mParamCond.broadcast(); 1115 // do not lock the mutex in destructor 1116 releaseWakeLock_l(); 1117 if (mPowerManager != 0) { 1118 sp<IBinder> binder = mPowerManager->asBinder(); 1119 binder->unlinkToDeath(mDeathRecipient); 1120 } 1121} 1122 1123void AudioFlinger::ThreadBase::exit() 1124{ 1125 ALOGV("ThreadBase::exit"); 1126 { 1127 // This lock prevents the following race in thread (uniprocessor for illustration): 1128 // if (!exitPending()) { 1129 // // context switch from here to exit() 1130 // // exit() calls requestExit(), what exitPending() observes 1131 // // exit() calls signal(), which is dropped since no waiters 1132 // // context switch back from exit() to here 1133 // mWaitWorkCV.wait(...); 1134 // // now thread is hung 1135 // } 1136 AutoMutex lock(mLock); 1137 requestExit(); 1138 mWaitWorkCV.signal(); 1139 } 1140 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1141 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1142 requestExitAndWait(); 1143} 1144 1145status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1146{ 1147 status_t status; 1148 1149 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1150 Mutex::Autolock _l(mLock); 1151 1152 mNewParameters.add(keyValuePairs); 1153 mWaitWorkCV.signal(); 1154 // wait condition with timeout in case the thread loop has exited 1155 // before the request could be processed 1156 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1157 status = mParamStatus; 1158 mWaitWorkCV.signal(); 1159 } else { 1160 status = TIMED_OUT; 1161 } 1162 return status; 1163} 1164 1165void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1166{ 1167 Mutex::Autolock _l(mLock); 1168 sendConfigEvent_l(event, param); 1169} 1170 1171// sendConfigEvent_l() must be called with ThreadBase::mLock held 1172void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1173{ 1174 ConfigEvent configEvent; 1175 configEvent.mEvent = event; 1176 configEvent.mParam = param; 1177 mConfigEvents.add(configEvent); 1178 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1179 mWaitWorkCV.signal(); 1180} 1181 1182void AudioFlinger::ThreadBase::processConfigEvents() 1183{ 1184 mLock.lock(); 1185 while (!mConfigEvents.isEmpty()) { 1186 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1187 ConfigEvent configEvent = mConfigEvents[0]; 1188 mConfigEvents.removeAt(0); 1189 // release mLock before locking AudioFlinger mLock: lock order is always 1190 // AudioFlinger then ThreadBase to avoid cross deadlock 1191 mLock.unlock(); 1192 mAudioFlinger->mLock.lock(); 1193 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1194 mAudioFlinger->mLock.unlock(); 1195 mLock.lock(); 1196 } 1197 mLock.unlock(); 1198} 1199 1200status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1201{ 1202 const size_t SIZE = 256; 1203 char buffer[SIZE]; 1204 String8 result; 1205 1206 bool locked = tryLock(mLock); 1207 if (!locked) { 1208 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1209 write(fd, buffer, strlen(buffer)); 1210 } 1211 1212 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1213 result.append(buffer); 1214 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1217 result.append(buffer); 1218 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1219 result.append(buffer); 1220 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1221 result.append(buffer); 1222 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1223 result.append(buffer); 1224 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1225 result.append(buffer); 1226 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1227 result.append(buffer); 1228 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1229 result.append(buffer); 1230 1231 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1232 result.append(buffer); 1233 result.append(" Index Command"); 1234 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1235 snprintf(buffer, SIZE, "\n %02d ", i); 1236 result.append(buffer); 1237 result.append(mNewParameters[i]); 1238 } 1239 1240 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1241 result.append(buffer); 1242 snprintf(buffer, SIZE, " Index event param\n"); 1243 result.append(buffer); 1244 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1245 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1246 result.append(buffer); 1247 } 1248 result.append("\n"); 1249 1250 write(fd, result.string(), result.size()); 1251 1252 if (locked) { 1253 mLock.unlock(); 1254 } 1255 return NO_ERROR; 1256} 1257 1258status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1259{ 1260 const size_t SIZE = 256; 1261 char buffer[SIZE]; 1262 String8 result; 1263 1264 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1265 write(fd, buffer, strlen(buffer)); 1266 1267 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1268 sp<EffectChain> chain = mEffectChains[i]; 1269 if (chain != 0) { 1270 chain->dump(fd, args); 1271 } 1272 } 1273 return NO_ERROR; 1274} 1275 1276void AudioFlinger::ThreadBase::acquireWakeLock() 1277{ 1278 Mutex::Autolock _l(mLock); 1279 acquireWakeLock_l(); 1280} 1281 1282void AudioFlinger::ThreadBase::acquireWakeLock_l() 1283{ 1284 if (mPowerManager == 0) { 1285 // use checkService() to avoid blocking if power service is not up yet 1286 sp<IBinder> binder = 1287 defaultServiceManager()->checkService(String16("power")); 1288 if (binder == 0) { 1289 ALOGW("Thread %s cannot connect to the power manager service", mName); 1290 } else { 1291 mPowerManager = interface_cast<IPowerManager>(binder); 1292 binder->linkToDeath(mDeathRecipient); 1293 } 1294 } 1295 if (mPowerManager != 0) { 1296 sp<IBinder> binder = new BBinder(); 1297 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1298 binder, 1299 String16(mName)); 1300 if (status == NO_ERROR) { 1301 mWakeLockToken = binder; 1302 } 1303 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1304 } 1305} 1306 1307void AudioFlinger::ThreadBase::releaseWakeLock() 1308{ 1309 Mutex::Autolock _l(mLock); 1310 releaseWakeLock_l(); 1311} 1312 1313void AudioFlinger::ThreadBase::releaseWakeLock_l() 1314{ 1315 if (mWakeLockToken != 0) { 1316 ALOGV("releaseWakeLock_l() %s", mName); 1317 if (mPowerManager != 0) { 1318 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1319 } 1320 mWakeLockToken.clear(); 1321 } 1322} 1323 1324void AudioFlinger::ThreadBase::clearPowerManager() 1325{ 1326 Mutex::Autolock _l(mLock); 1327 releaseWakeLock_l(); 1328 mPowerManager.clear(); 1329} 1330 1331void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1332{ 1333 sp<ThreadBase> thread = mThread.promote(); 1334 if (thread != 0) { 1335 thread->clearPowerManager(); 1336 } 1337 ALOGW("power manager service died !!!"); 1338} 1339 1340void AudioFlinger::ThreadBase::setEffectSuspended( 1341 const effect_uuid_t *type, bool suspend, int sessionId) 1342{ 1343 Mutex::Autolock _l(mLock); 1344 setEffectSuspended_l(type, suspend, sessionId); 1345} 1346 1347void AudioFlinger::ThreadBase::setEffectSuspended_l( 1348 const effect_uuid_t *type, bool suspend, int sessionId) 1349{ 1350 sp<EffectChain> chain = getEffectChain_l(sessionId); 1351 if (chain != 0) { 1352 if (type != NULL) { 1353 chain->setEffectSuspended_l(type, suspend); 1354 } else { 1355 chain->setEffectSuspendedAll_l(suspend); 1356 } 1357 } 1358 1359 updateSuspendedSessions_l(type, suspend, sessionId); 1360} 1361 1362void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1363{ 1364 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1365 if (index < 0) { 1366 return; 1367 } 1368 1369 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1370 mSuspendedSessions.editValueAt(index); 1371 1372 for (size_t i = 0; i < sessionEffects.size(); i++) { 1373 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1374 for (int j = 0; j < desc->mRefCount; j++) { 1375 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1376 chain->setEffectSuspendedAll_l(true); 1377 } else { 1378 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1379 desc->mType.timeLow); 1380 chain->setEffectSuspended_l(&desc->mType, true); 1381 } 1382 } 1383 } 1384} 1385 1386void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1387 bool suspend, 1388 int sessionId) 1389{ 1390 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1391 1392 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1393 1394 if (suspend) { 1395 if (index >= 0) { 1396 sessionEffects = mSuspendedSessions.editValueAt(index); 1397 } else { 1398 mSuspendedSessions.add(sessionId, sessionEffects); 1399 } 1400 } else { 1401 if (index < 0) { 1402 return; 1403 } 1404 sessionEffects = mSuspendedSessions.editValueAt(index); 1405 } 1406 1407 1408 int key = EffectChain::kKeyForSuspendAll; 1409 if (type != NULL) { 1410 key = type->timeLow; 1411 } 1412 index = sessionEffects.indexOfKey(key); 1413 1414 sp<SuspendedSessionDesc> desc; 1415 if (suspend) { 1416 if (index >= 0) { 1417 desc = sessionEffects.valueAt(index); 1418 } else { 1419 desc = new SuspendedSessionDesc(); 1420 if (type != NULL) { 1421 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1422 } 1423 sessionEffects.add(key, desc); 1424 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1425 } 1426 desc->mRefCount++; 1427 } else { 1428 if (index < 0) { 1429 return; 1430 } 1431 desc = sessionEffects.valueAt(index); 1432 if (--desc->mRefCount == 0) { 1433 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1434 sessionEffects.removeItemsAt(index); 1435 if (sessionEffects.isEmpty()) { 1436 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1437 sessionId); 1438 mSuspendedSessions.removeItem(sessionId); 1439 } 1440 } 1441 } 1442 if (!sessionEffects.isEmpty()) { 1443 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1444 } 1445} 1446 1447void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1448 bool enabled, 1449 int sessionId) 1450{ 1451 Mutex::Autolock _l(mLock); 1452 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1453} 1454 1455void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1456 bool enabled, 1457 int sessionId) 1458{ 1459 if (mType != RECORD) { 1460 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1461 // another session. This gives the priority to well behaved effect control panels 1462 // and applications not using global effects. 1463 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1464 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1465 } 1466 } 1467 1468 sp<EffectChain> chain = getEffectChain_l(sessionId); 1469 if (chain != 0) { 1470 chain->checkSuspendOnEffectEnabled(effect, enabled); 1471 } 1472} 1473 1474// ---------------------------------------------------------------------------- 1475 1476AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1477 AudioStreamOut* output, 1478 audio_io_handle_t id, 1479 uint32_t device, 1480 type_t type) 1481 : ThreadBase(audioFlinger, id, device, type), 1482 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1483 // Assumes constructor is called by AudioFlinger with it's mLock held, 1484 // but it would be safer to explicitly pass initial masterMute as parameter 1485 mMasterMute(audioFlinger->masterMute_l()), 1486 // mStreamTypes[] initialized in constructor body 1487 mOutput(output), 1488 // Assumes constructor is called by AudioFlinger with it's mLock held, 1489 // but it would be safer to explicitly pass initial masterVolume as parameter 1490 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1491 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1492 mMixerStatus(MIXER_IDLE), 1493 mPrevMixerStatus(MIXER_IDLE), 1494 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1495{ 1496 snprintf(mName, kNameLength, "AudioOut_%X", id); 1497 1498 readOutputParameters(); 1499 1500 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1501 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1502 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1503 stream = (audio_stream_type_t) (stream + 1)) { 1504 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1505 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1506 // initialized by stream_type_t default constructor 1507 // mStreamTypes[stream].valid = true; 1508 } 1509 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1510 // because mAudioFlinger doesn't have one to copy from 1511} 1512 1513AudioFlinger::PlaybackThread::~PlaybackThread() 1514{ 1515 delete [] mMixBuffer; 1516} 1517 1518status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1519{ 1520 dumpInternals(fd, args); 1521 dumpTracks(fd, args); 1522 dumpEffectChains(fd, args); 1523 return NO_ERROR; 1524} 1525 1526status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1527{ 1528 const size_t SIZE = 256; 1529 char buffer[SIZE]; 1530 String8 result; 1531 1532 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1533 result.append(buffer); 1534 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1535 for (size_t i = 0; i < mTracks.size(); ++i) { 1536 sp<Track> track = mTracks[i]; 1537 if (track != 0) { 1538 track->dump(buffer, SIZE); 1539 result.append(buffer); 1540 } 1541 } 1542 1543 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1544 result.append(buffer); 1545 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1546 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1547 sp<Track> track = mActiveTracks[i].promote(); 1548 if (track != 0) { 1549 track->dump(buffer, SIZE); 1550 result.append(buffer); 1551 } 1552 } 1553 write(fd, result.string(), result.size()); 1554 return NO_ERROR; 1555} 1556 1557status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1558{ 1559 const size_t SIZE = 256; 1560 char buffer[SIZE]; 1561 String8 result; 1562 1563 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1564 result.append(buffer); 1565 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1566 result.append(buffer); 1567 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1568 result.append(buffer); 1569 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1570 result.append(buffer); 1571 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1572 result.append(buffer); 1573 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1574 result.append(buffer); 1575 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1576 result.append(buffer); 1577 write(fd, result.string(), result.size()); 1578 1579 dumpBase(fd, args); 1580 1581 return NO_ERROR; 1582} 1583 1584// Thread virtuals 1585status_t AudioFlinger::PlaybackThread::readyToRun() 1586{ 1587 status_t status = initCheck(); 1588 if (status == NO_ERROR) { 1589 ALOGI("AudioFlinger's thread %p ready to run", this); 1590 } else { 1591 ALOGE("No working audio driver found."); 1592 } 1593 return status; 1594} 1595 1596void AudioFlinger::PlaybackThread::onFirstRef() 1597{ 1598 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1599} 1600 1601// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1602sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1603 const sp<AudioFlinger::Client>& client, 1604 audio_stream_type_t streamType, 1605 uint32_t sampleRate, 1606 audio_format_t format, 1607 uint32_t channelMask, 1608 int frameCount, 1609 const sp<IMemory>& sharedBuffer, 1610 int sessionId, 1611 bool isTimed, 1612 status_t *status) 1613{ 1614 sp<Track> track; 1615 status_t lStatus; 1616 1617 if (mType == DIRECT) { 1618 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1619 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1620 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1621 "for output %p with format %d", 1622 sampleRate, format, channelMask, mOutput, mFormat); 1623 lStatus = BAD_VALUE; 1624 goto Exit; 1625 } 1626 } 1627 } else { 1628 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1629 if (sampleRate > mSampleRate*2) { 1630 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1631 lStatus = BAD_VALUE; 1632 goto Exit; 1633 } 1634 } 1635 1636 lStatus = initCheck(); 1637 if (lStatus != NO_ERROR) { 1638 ALOGE("Audio driver not initialized."); 1639 goto Exit; 1640 } 1641 1642 { // scope for mLock 1643 Mutex::Autolock _l(mLock); 1644 1645 // all tracks in same audio session must share the same routing strategy otherwise 1646 // conflicts will happen when tracks are moved from one output to another by audio policy 1647 // manager 1648 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1649 for (size_t i = 0; i < mTracks.size(); ++i) { 1650 sp<Track> t = mTracks[i]; 1651 if (t != 0 && !t->isOutputTrack()) { 1652 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1653 if (sessionId == t->sessionId() && strategy != actual) { 1654 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1655 strategy, actual); 1656 lStatus = BAD_VALUE; 1657 goto Exit; 1658 } 1659 } 1660 } 1661 1662 if (!isTimed) { 1663 track = new Track(this, client, streamType, sampleRate, format, 1664 channelMask, frameCount, sharedBuffer, sessionId); 1665 } else { 1666 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1667 channelMask, frameCount, sharedBuffer, sessionId); 1668 } 1669 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1670 lStatus = NO_MEMORY; 1671 goto Exit; 1672 } 1673 mTracks.add(track); 1674 1675 sp<EffectChain> chain = getEffectChain_l(sessionId); 1676 if (chain != 0) { 1677 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1678 track->setMainBuffer(chain->inBuffer()); 1679 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1680 chain->incTrackCnt(); 1681 } 1682 1683 // invalidate track immediately if the stream type was moved to another thread since 1684 // createTrack() was called by the client process. 1685 if (!mStreamTypes[streamType].valid) { 1686 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1687 this, streamType); 1688 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1689 } 1690 } 1691 lStatus = NO_ERROR; 1692 1693Exit: 1694 if (status) { 1695 *status = lStatus; 1696 } 1697 return track; 1698} 1699 1700uint32_t AudioFlinger::PlaybackThread::latency() const 1701{ 1702 Mutex::Autolock _l(mLock); 1703 if (initCheck() == NO_ERROR) { 1704 return mOutput->stream->get_latency(mOutput->stream); 1705 } else { 1706 return 0; 1707 } 1708} 1709 1710void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1711{ 1712 Mutex::Autolock _l(mLock); 1713 mMasterVolume = value; 1714} 1715 1716void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1717{ 1718 Mutex::Autolock _l(mLock); 1719 setMasterMute_l(muted); 1720} 1721 1722void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1723{ 1724 Mutex::Autolock _l(mLock); 1725 mStreamTypes[stream].volume = value; 1726} 1727 1728void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1729{ 1730 Mutex::Autolock _l(mLock); 1731 mStreamTypes[stream].mute = muted; 1732} 1733 1734float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1735{ 1736 Mutex::Autolock _l(mLock); 1737 return mStreamTypes[stream].volume; 1738} 1739 1740// addTrack_l() must be called with ThreadBase::mLock held 1741status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1742{ 1743 status_t status = ALREADY_EXISTS; 1744 1745 // set retry count for buffer fill 1746 track->mRetryCount = kMaxTrackStartupRetries; 1747 if (mActiveTracks.indexOf(track) < 0) { 1748 // the track is newly added, make sure it fills up all its 1749 // buffers before playing. This is to ensure the client will 1750 // effectively get the latency it requested. 1751 track->mFillingUpStatus = Track::FS_FILLING; 1752 track->mResetDone = false; 1753 mActiveTracks.add(track); 1754 if (track->mainBuffer() != mMixBuffer) { 1755 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1756 if (chain != 0) { 1757 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1758 chain->incActiveTrackCnt(); 1759 } 1760 } 1761 1762 status = NO_ERROR; 1763 } 1764 1765 ALOGV("mWaitWorkCV.broadcast"); 1766 mWaitWorkCV.broadcast(); 1767 1768 return status; 1769} 1770 1771// destroyTrack_l() must be called with ThreadBase::mLock held 1772void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1773{ 1774 track->mState = TrackBase::TERMINATED; 1775 if (mActiveTracks.indexOf(track) < 0) { 1776 removeTrack_l(track); 1777 } 1778} 1779 1780void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1781{ 1782 mTracks.remove(track); 1783 deleteTrackName_l(track->name()); 1784 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1785 if (chain != 0) { 1786 chain->decTrackCnt(); 1787 } 1788} 1789 1790String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1791{ 1792 String8 out_s8 = String8(""); 1793 char *s; 1794 1795 Mutex::Autolock _l(mLock); 1796 if (initCheck() != NO_ERROR) { 1797 return out_s8; 1798 } 1799 1800 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1801 out_s8 = String8(s); 1802 free(s); 1803 return out_s8; 1804} 1805 1806// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1807void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1808 AudioSystem::OutputDescriptor desc; 1809 void *param2 = NULL; 1810 1811 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1812 1813 switch (event) { 1814 case AudioSystem::OUTPUT_OPENED: 1815 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1816 desc.channels = mChannelMask; 1817 desc.samplingRate = mSampleRate; 1818 desc.format = mFormat; 1819 desc.frameCount = mFrameCount; 1820 desc.latency = latency(); 1821 param2 = &desc; 1822 break; 1823 1824 case AudioSystem::STREAM_CONFIG_CHANGED: 1825 param2 = ¶m; 1826 case AudioSystem::OUTPUT_CLOSED: 1827 default: 1828 break; 1829 } 1830 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1831} 1832 1833void AudioFlinger::PlaybackThread::readOutputParameters() 1834{ 1835 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1836 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1837 mChannelCount = (uint16_t)popcount(mChannelMask); 1838 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1839 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1840 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1841 1842 // FIXME - Current mixer implementation only supports stereo output: Always 1843 // Allocate a stereo buffer even if HW output is mono. 1844 delete[] mMixBuffer; 1845 mMixBuffer = new int16_t[mFrameCount * 2]; 1846 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1847 1848 // force reconfiguration of effect chains and engines to take new buffer size and audio 1849 // parameters into account 1850 // Note that mLock is not held when readOutputParameters() is called from the constructor 1851 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1852 // matter. 1853 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1854 Vector< sp<EffectChain> > effectChains = mEffectChains; 1855 for (size_t i = 0; i < effectChains.size(); i ++) { 1856 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1857 } 1858} 1859 1860status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1861{ 1862 if (halFrames == NULL || dspFrames == NULL) { 1863 return BAD_VALUE; 1864 } 1865 Mutex::Autolock _l(mLock); 1866 if (initCheck() != NO_ERROR) { 1867 return INVALID_OPERATION; 1868 } 1869 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1870 1871 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1872} 1873 1874uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1875{ 1876 Mutex::Autolock _l(mLock); 1877 uint32_t result = 0; 1878 if (getEffectChain_l(sessionId) != 0) { 1879 result = EFFECT_SESSION; 1880 } 1881 1882 for (size_t i = 0; i < mTracks.size(); ++i) { 1883 sp<Track> track = mTracks[i]; 1884 if (sessionId == track->sessionId() && 1885 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1886 result |= TRACK_SESSION; 1887 break; 1888 } 1889 } 1890 1891 return result; 1892} 1893 1894uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1895{ 1896 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1897 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1898 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1899 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1900 } 1901 for (size_t i = 0; i < mTracks.size(); i++) { 1902 sp<Track> track = mTracks[i]; 1903 if (sessionId == track->sessionId() && 1904 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1905 return AudioSystem::getStrategyForStream(track->streamType()); 1906 } 1907 } 1908 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1909} 1910 1911 1912AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1913{ 1914 Mutex::Autolock _l(mLock); 1915 return mOutput; 1916} 1917 1918AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1919{ 1920 Mutex::Autolock _l(mLock); 1921 AudioStreamOut *output = mOutput; 1922 mOutput = NULL; 1923 return output; 1924} 1925 1926// this method must always be called either with ThreadBase mLock held or inside the thread loop 1927audio_stream_t* AudioFlinger::PlaybackThread::stream() 1928{ 1929 if (mOutput == NULL) { 1930 return NULL; 1931 } 1932 return &mOutput->stream->common; 1933} 1934 1935uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1936{ 1937 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1938 // decoding and transfer time. So sleeping for half of the latency would likely cause 1939 // underruns 1940 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1941 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1942 } else { 1943 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1944 } 1945} 1946 1947status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1948{ 1949 if (!isValidSyncEvent(event)) { 1950 return BAD_VALUE; 1951 } 1952 1953 Mutex::Autolock _l(mLock); 1954 1955 for (size_t i = 0; i < mTracks.size(); ++i) { 1956 sp<Track> track = mTracks[i]; 1957 if (event->triggerSession() == track->sessionId()) { 1958 track->setSyncEvent(event); 1959 return NO_ERROR; 1960 } 1961 } 1962 1963 return NAME_NOT_FOUND; 1964} 1965 1966bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 1967{ 1968 switch (event->type()) { 1969 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 1970 return true; 1971 default: 1972 break; 1973 } 1974 return false; 1975} 1976 1977// ---------------------------------------------------------------------------- 1978 1979AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1980 audio_io_handle_t id, uint32_t device, type_t type) 1981 : PlaybackThread(audioFlinger, output, id, device, type) 1982{ 1983 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1984 // FIXME - Current mixer implementation only supports stereo output 1985 if (mChannelCount == 1) { 1986 ALOGE("Invalid audio hardware channel count"); 1987 } 1988} 1989 1990AudioFlinger::MixerThread::~MixerThread() 1991{ 1992 delete mAudioMixer; 1993} 1994 1995class CpuStats { 1996public: 1997 CpuStats(); 1998 void sample(const String8 &title); 1999#ifdef DEBUG_CPU_USAGE 2000private: 2001 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2002 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2003 2004 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2005 2006 int mCpuNum; // thread's current CPU number 2007 int mCpukHz; // frequency of thread's current CPU in kHz 2008#endif 2009}; 2010 2011CpuStats::CpuStats() 2012#ifdef DEBUG_CPU_USAGE 2013 : mCpuNum(-1), mCpukHz(-1) 2014#endif 2015{ 2016} 2017 2018void CpuStats::sample(const String8 &title) { 2019#ifdef DEBUG_CPU_USAGE 2020 // get current thread's delta CPU time in wall clock ns 2021 double wcNs; 2022 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2023 2024 // record sample for wall clock statistics 2025 if (valid) { 2026 mWcStats.sample(wcNs); 2027 } 2028 2029 // get the current CPU number 2030 int cpuNum = sched_getcpu(); 2031 2032 // get the current CPU frequency in kHz 2033 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2034 2035 // check if either CPU number or frequency changed 2036 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2037 mCpuNum = cpuNum; 2038 mCpukHz = cpukHz; 2039 // ignore sample for purposes of cycles 2040 valid = false; 2041 } 2042 2043 // if no change in CPU number or frequency, then record sample for cycle statistics 2044 if (valid && mCpukHz > 0) { 2045 double cycles = wcNs * cpukHz * 0.000001; 2046 mHzStats.sample(cycles); 2047 } 2048 2049 unsigned n = mWcStats.n(); 2050 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2051 if ((n & 127) == 1) { 2052 long long elapsed = mCpuUsage.elapsed(); 2053 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2054 double perLoop = elapsed / (double) n; 2055 double perLoop100 = perLoop * 0.01; 2056 double perLoop1k = perLoop * 0.001; 2057 double mean = mWcStats.mean(); 2058 double stddev = mWcStats.stddev(); 2059 double minimum = mWcStats.minimum(); 2060 double maximum = mWcStats.maximum(); 2061 double meanCycles = mHzStats.mean(); 2062 double stddevCycles = mHzStats.stddev(); 2063 double minCycles = mHzStats.minimum(); 2064 double maxCycles = mHzStats.maximum(); 2065 mCpuUsage.resetElapsed(); 2066 mWcStats.reset(); 2067 mHzStats.reset(); 2068 ALOGD("CPU usage for %s over past %.1f secs\n" 2069 " (%u mixer loops at %.1f mean ms per loop):\n" 2070 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2071 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2072 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2073 title.string(), 2074 elapsed * .000000001, n, perLoop * .000001, 2075 mean * .001, 2076 stddev * .001, 2077 minimum * .001, 2078 maximum * .001, 2079 mean / perLoop100, 2080 stddev / perLoop100, 2081 minimum / perLoop100, 2082 maximum / perLoop100, 2083 meanCycles / perLoop1k, 2084 stddevCycles / perLoop1k, 2085 minCycles / perLoop1k, 2086 maxCycles / perLoop1k); 2087 2088 } 2089 } 2090#endif 2091}; 2092 2093void AudioFlinger::PlaybackThread::checkSilentMode_l() 2094{ 2095 if (!mMasterMute) { 2096 char value[PROPERTY_VALUE_MAX]; 2097 if (property_get("ro.audio.silent", value, "0") > 0) { 2098 char *endptr; 2099 unsigned long ul = strtoul(value, &endptr, 0); 2100 if (*endptr == '\0' && ul != 0) { 2101 ALOGD("Silence is golden"); 2102 // The setprop command will not allow a property to be changed after 2103 // the first time it is set, so we don't have to worry about un-muting. 2104 setMasterMute_l(true); 2105 } 2106 } 2107 } 2108} 2109 2110bool AudioFlinger::PlaybackThread::threadLoop() 2111{ 2112 Vector< sp<Track> > tracksToRemove; 2113 2114 standbyTime = systemTime(); 2115 2116 // MIXER 2117 nsecs_t lastWarning = 0; 2118if (mType == MIXER) { 2119 longStandbyExit = false; 2120} 2121 2122 // DUPLICATING 2123 // FIXME could this be made local to while loop? 2124 writeFrames = 0; 2125 2126 cacheParameters_l(); 2127 sleepTime = idleSleepTime; 2128 2129if (mType == MIXER) { 2130 sleepTimeShift = 0; 2131} 2132 2133 CpuStats cpuStats; 2134 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2135 2136 acquireWakeLock(); 2137 2138 while (!exitPending()) 2139 { 2140 cpuStats.sample(myName); 2141 2142 Vector< sp<EffectChain> > effectChains; 2143 2144 processConfigEvents(); 2145 2146 { // scope for mLock 2147 2148 Mutex::Autolock _l(mLock); 2149 2150 if (checkForNewParameters_l()) { 2151 cacheParameters_l(); 2152 } 2153 2154 saveOutputTracks(); 2155 2156 // put audio hardware into standby after short delay 2157 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2158 mSuspended > 0)) { 2159 if (!mStandby) { 2160 2161 threadLoop_standby(); 2162 2163 mStandby = true; 2164 mBytesWritten = 0; 2165 } 2166 2167 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2168 // we're about to wait, flush the binder command buffer 2169 IPCThreadState::self()->flushCommands(); 2170 2171 clearOutputTracks(); 2172 2173 if (exitPending()) break; 2174 2175 releaseWakeLock_l(); 2176 // wait until we have something to do... 2177 ALOGV("%s going to sleep", myName.string()); 2178 mWaitWorkCV.wait(mLock); 2179 ALOGV("%s waking up", myName.string()); 2180 acquireWakeLock_l(); 2181 2182 mPrevMixerStatus = MIXER_IDLE; 2183 2184 checkSilentMode_l(); 2185 2186 standbyTime = systemTime() + standbyDelay; 2187 sleepTime = idleSleepTime; 2188 if (mType == MIXER) { 2189 sleepTimeShift = 0; 2190 } 2191 2192 continue; 2193 } 2194 } 2195 2196 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2197 // Shift in the new status; this could be a queue if it's 2198 // useful to filter the mixer status over several cycles. 2199 mPrevMixerStatus = mMixerStatus; 2200 mMixerStatus = newMixerStatus; 2201 2202 // prevent any changes in effect chain list and in each effect chain 2203 // during mixing and effect process as the audio buffers could be deleted 2204 // or modified if an effect is created or deleted 2205 lockEffectChains_l(effectChains); 2206 } 2207 2208 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2209 threadLoop_mix(); 2210 } else { 2211 threadLoop_sleepTime(); 2212 } 2213 2214 if (mSuspended > 0) { 2215 sleepTime = suspendSleepTimeUs(); 2216 } 2217 2218 // only process effects if we're going to write 2219 if (sleepTime == 0) { 2220 for (size_t i = 0; i < effectChains.size(); i ++) { 2221 effectChains[i]->process_l(); 2222 } 2223 } 2224 2225 // enable changes in effect chain 2226 unlockEffectChains(effectChains); 2227 2228 // sleepTime == 0 means we must write to audio hardware 2229 if (sleepTime == 0) { 2230 2231 threadLoop_write(); 2232 2233if (mType == MIXER) { 2234 // write blocked detection 2235 nsecs_t now = systemTime(); 2236 nsecs_t delta = now - mLastWriteTime; 2237 if (!mStandby && delta > maxPeriod) { 2238 mNumDelayedWrites++; 2239 if ((now - lastWarning) > kWarningThrottleNs) { 2240 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2241 ns2ms(delta), mNumDelayedWrites, this); 2242 lastWarning = now; 2243 } 2244 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2245 // a different threshold. Or completely removed for what it is worth anyway... 2246 if (mStandby) { 2247 longStandbyExit = true; 2248 } 2249 } 2250} 2251 2252 mStandby = false; 2253 } else { 2254 usleep(sleepTime); 2255 } 2256 2257 // finally let go of removed track(s), without the lock held 2258 // since we can't guarantee the destructors won't acquire that 2259 // same lock. 2260 tracksToRemove.clear(); 2261 2262 // FIXME I don't understand the need for this here; 2263 // it was in the original code but maybe the 2264 // assignment in saveOutputTracks() makes this unnecessary? 2265 clearOutputTracks(); 2266 2267 // Effect chains will be actually deleted here if they were removed from 2268 // mEffectChains list during mixing or effects processing 2269 effectChains.clear(); 2270 2271 // FIXME Note that the above .clear() is no longer necessary since effectChains 2272 // is now local to this block, but will keep it for now (at least until merge done). 2273 } 2274 2275if (mType == MIXER || mType == DIRECT) { 2276 // put output stream into standby mode 2277 if (!mStandby) { 2278 mOutput->stream->common.standby(&mOutput->stream->common); 2279 } 2280} 2281if (mType == DUPLICATING) { 2282 // for DuplicatingThread, standby mode is handled by the outputTracks 2283} 2284 2285 releaseWakeLock(); 2286 2287 ALOGV("Thread %p type %d exiting", this, mType); 2288 return false; 2289} 2290 2291// shared by MIXER and DIRECT, overridden by DUPLICATING 2292void AudioFlinger::PlaybackThread::threadLoop_write() 2293{ 2294 // FIXME rewrite to reduce number of system calls 2295 mLastWriteTime = systemTime(); 2296 mInWrite = true; 2297 mBytesWritten += mixBufferSize; 2298 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2299 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2300 mNumWrites++; 2301 mInWrite = false; 2302} 2303 2304// shared by MIXER and DIRECT, overridden by DUPLICATING 2305void AudioFlinger::PlaybackThread::threadLoop_standby() 2306{ 2307 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2308 mOutput->stream->common.standby(&mOutput->stream->common); 2309} 2310 2311void AudioFlinger::MixerThread::threadLoop_mix() 2312{ 2313 // obtain the presentation timestamp of the next output buffer 2314 int64_t pts; 2315 status_t status = INVALID_OPERATION; 2316 2317 if (NULL != mOutput->stream->get_next_write_timestamp) { 2318 status = mOutput->stream->get_next_write_timestamp( 2319 mOutput->stream, &pts); 2320 } 2321 2322 if (status != NO_ERROR) { 2323 pts = AudioBufferProvider::kInvalidPTS; 2324 } 2325 2326 // mix buffers... 2327 mAudioMixer->process(pts); 2328 // increase sleep time progressively when application underrun condition clears. 2329 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2330 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2331 // such that we would underrun the audio HAL. 2332 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2333 sleepTimeShift--; 2334 } 2335 sleepTime = 0; 2336 standbyTime = systemTime() + standbyDelay; 2337 //TODO: delay standby when effects have a tail 2338} 2339 2340void AudioFlinger::MixerThread::threadLoop_sleepTime() 2341{ 2342 // If no tracks are ready, sleep once for the duration of an output 2343 // buffer size, then write 0s to the output 2344 if (sleepTime == 0) { 2345 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2346 sleepTime = activeSleepTime >> sleepTimeShift; 2347 if (sleepTime < kMinThreadSleepTimeUs) { 2348 sleepTime = kMinThreadSleepTimeUs; 2349 } 2350 // reduce sleep time in case of consecutive application underruns to avoid 2351 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2352 // duration we would end up writing less data than needed by the audio HAL if 2353 // the condition persists. 2354 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2355 sleepTimeShift++; 2356 } 2357 } else { 2358 sleepTime = idleSleepTime; 2359 } 2360 } else if (mBytesWritten != 0 || 2361 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2362 memset (mMixBuffer, 0, mixBufferSize); 2363 sleepTime = 0; 2364 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2365 } 2366 // TODO add standby time extension fct of effect tail 2367} 2368 2369// prepareTracks_l() must be called with ThreadBase::mLock held 2370AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2371 Vector< sp<Track> > *tracksToRemove) 2372{ 2373 2374 mixer_state mixerStatus = MIXER_IDLE; 2375 // find out which tracks need to be processed 2376 size_t count = mActiveTracks.size(); 2377 size_t mixedTracks = 0; 2378 size_t tracksWithEffect = 0; 2379 2380 float masterVolume = mMasterVolume; 2381 bool masterMute = mMasterMute; 2382 2383 if (masterMute) { 2384 masterVolume = 0; 2385 } 2386 // Delegate master volume control to effect in output mix effect chain if needed 2387 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2388 if (chain != 0) { 2389 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2390 chain->setVolume_l(&v, &v); 2391 masterVolume = (float)((v + (1 << 23)) >> 24); 2392 chain.clear(); 2393 } 2394 2395 for (size_t i=0 ; i<count ; i++) { 2396 sp<Track> t = mActiveTracks[i].promote(); 2397 if (t == 0) continue; 2398 2399 // this const just means the local variable doesn't change 2400 Track* const track = t.get(); 2401 audio_track_cblk_t* cblk = track->cblk(); 2402 2403 // The first time a track is added we wait 2404 // for all its buffers to be filled before processing it 2405 int name = track->name(); 2406 // make sure that we have enough frames to mix one full buffer. 2407 // enforce this condition only once to enable draining the buffer in case the client 2408 // app does not call stop() and relies on underrun to stop: 2409 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2410 // during last round 2411 uint32_t minFrames = 1; 2412 if (!track->isStopped() && !track->isPausing() && 2413 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2414 if (t->sampleRate() == (int)mSampleRate) { 2415 minFrames = mFrameCount; 2416 } else { 2417 // +1 for rounding and +1 for additional sample needed for interpolation 2418 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2419 // add frames already consumed but not yet released by the resampler 2420 // because cblk->framesReady() will include these frames 2421 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2422 // the minimum track buffer size is normally twice the number of frames necessary 2423 // to fill one buffer and the resampler should not leave more than one buffer worth 2424 // of unreleased frames after each pass, but just in case... 2425 ALOG_ASSERT(minFrames <= cblk->frameCount); 2426 } 2427 } 2428 if ((track->framesReady() >= minFrames) && track->isReady() && 2429 !track->isPaused() && !track->isTerminated()) 2430 { 2431 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2432 2433 mixedTracks++; 2434 2435 // track->mainBuffer() != mMixBuffer means there is an effect chain 2436 // connected to the track 2437 chain.clear(); 2438 if (track->mainBuffer() != mMixBuffer) { 2439 chain = getEffectChain_l(track->sessionId()); 2440 // Delegate volume control to effect in track effect chain if needed 2441 if (chain != 0) { 2442 tracksWithEffect++; 2443 } else { 2444 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2445 name, track->sessionId()); 2446 } 2447 } 2448 2449 2450 int param = AudioMixer::VOLUME; 2451 if (track->mFillingUpStatus == Track::FS_FILLED) { 2452 // no ramp for the first volume setting 2453 track->mFillingUpStatus = Track::FS_ACTIVE; 2454 if (track->mState == TrackBase::RESUMING) { 2455 track->mState = TrackBase::ACTIVE; 2456 param = AudioMixer::RAMP_VOLUME; 2457 } 2458 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2459 } else if (cblk->server != 0) { 2460 // If the track is stopped before the first frame was mixed, 2461 // do not apply ramp 2462 param = AudioMixer::RAMP_VOLUME; 2463 } 2464 2465 // compute volume for this track 2466 uint32_t vl, vr, va; 2467 if (track->isMuted() || track->isPausing() || 2468 mStreamTypes[track->streamType()].mute) { 2469 vl = vr = va = 0; 2470 if (track->isPausing()) { 2471 track->setPaused(); 2472 } 2473 } else { 2474 2475 // read original volumes with volume control 2476 float typeVolume = mStreamTypes[track->streamType()].volume; 2477 float v = masterVolume * typeVolume; 2478 uint32_t vlr = cblk->getVolumeLR(); 2479 vl = vlr & 0xFFFF; 2480 vr = vlr >> 16; 2481 // track volumes come from shared memory, so can't be trusted and must be clamped 2482 if (vl > MAX_GAIN_INT) { 2483 ALOGV("Track left volume out of range: %04X", vl); 2484 vl = MAX_GAIN_INT; 2485 } 2486 if (vr > MAX_GAIN_INT) { 2487 ALOGV("Track right volume out of range: %04X", vr); 2488 vr = MAX_GAIN_INT; 2489 } 2490 // now apply the master volume and stream type volume 2491 vl = (uint32_t)(v * vl) << 12; 2492 vr = (uint32_t)(v * vr) << 12; 2493 // assuming master volume and stream type volume each go up to 1.0, 2494 // vl and vr are now in 8.24 format 2495 2496 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2497 // send level comes from shared memory and so may be corrupt 2498 if (sendLevel > MAX_GAIN_INT) { 2499 ALOGV("Track send level out of range: %04X", sendLevel); 2500 sendLevel = MAX_GAIN_INT; 2501 } 2502 va = (uint32_t)(v * sendLevel); 2503 } 2504 // Delegate volume control to effect in track effect chain if needed 2505 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2506 // Do not ramp volume if volume is controlled by effect 2507 param = AudioMixer::VOLUME; 2508 track->mHasVolumeController = true; 2509 } else { 2510 // force no volume ramp when volume controller was just disabled or removed 2511 // from effect chain to avoid volume spike 2512 if (track->mHasVolumeController) { 2513 param = AudioMixer::VOLUME; 2514 } 2515 track->mHasVolumeController = false; 2516 } 2517 2518 // Convert volumes from 8.24 to 4.12 format 2519 // This additional clamping is needed in case chain->setVolume_l() overshot 2520 vl = (vl + (1 << 11)) >> 12; 2521 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2522 vr = (vr + (1 << 11)) >> 12; 2523 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2524 2525 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2526 2527 // XXX: these things DON'T need to be done each time 2528 mAudioMixer->setBufferProvider(name, track); 2529 mAudioMixer->enable(name); 2530 2531 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2532 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2533 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2534 mAudioMixer->setParameter( 2535 name, 2536 AudioMixer::TRACK, 2537 AudioMixer::FORMAT, (void *)track->format()); 2538 mAudioMixer->setParameter( 2539 name, 2540 AudioMixer::TRACK, 2541 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2542 mAudioMixer->setParameter( 2543 name, 2544 AudioMixer::RESAMPLE, 2545 AudioMixer::SAMPLE_RATE, 2546 (void *)(cblk->sampleRate)); 2547 mAudioMixer->setParameter( 2548 name, 2549 AudioMixer::TRACK, 2550 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2551 mAudioMixer->setParameter( 2552 name, 2553 AudioMixer::TRACK, 2554 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2555 2556 // reset retry count 2557 track->mRetryCount = kMaxTrackRetries; 2558 2559 // If one track is ready, set the mixer ready if: 2560 // - the mixer was not ready during previous round OR 2561 // - no other track is not ready 2562 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2563 mixerStatus != MIXER_TRACKS_ENABLED) { 2564 mixerStatus = MIXER_TRACKS_READY; 2565 } 2566 } else { 2567 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2568 if (track->isStopped()) { 2569 track->reset(); 2570 } 2571 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2572 // We have consumed all the buffers of this track. 2573 // Remove it from the list of active tracks. 2574 // TODO: use actual buffer filling status instead of latency when available from 2575 // audio HAL 2576 size_t audioHALFrames = 2577 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2578 size_t framesWritten = 2579 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2580 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2581 tracksToRemove->add(track); 2582 } 2583 } else { 2584 // No buffers for this track. Give it a few chances to 2585 // fill a buffer, then remove it from active list. 2586 if (--(track->mRetryCount) <= 0) { 2587 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2588 tracksToRemove->add(track); 2589 // indicate to client process that the track was disabled because of underrun 2590 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2591 // If one track is not ready, mark the mixer also not ready if: 2592 // - the mixer was ready during previous round OR 2593 // - no other track is ready 2594 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2595 mixerStatus != MIXER_TRACKS_READY) { 2596 mixerStatus = MIXER_TRACKS_ENABLED; 2597 } 2598 } 2599 mAudioMixer->disable(name); 2600 } 2601 } 2602 2603 // remove all the tracks that need to be... 2604 count = tracksToRemove->size(); 2605 if (CC_UNLIKELY(count)) { 2606 for (size_t i=0 ; i<count ; i++) { 2607 const sp<Track>& track = tracksToRemove->itemAt(i); 2608 mActiveTracks.remove(track); 2609 if (track->mainBuffer() != mMixBuffer) { 2610 chain = getEffectChain_l(track->sessionId()); 2611 if (chain != 0) { 2612 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2613 chain->decActiveTrackCnt(); 2614 } 2615 } 2616 if (track->isTerminated()) { 2617 removeTrack_l(track); 2618 } 2619 } 2620 } 2621 2622 // mix buffer must be cleared if all tracks are connected to an 2623 // effect chain as in this case the mixer will not write to 2624 // mix buffer and track effects will accumulate into it 2625 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2626 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2627 } 2628 2629 return mixerStatus; 2630} 2631 2632/* 2633The derived values that are cached: 2634 - mixBufferSize from frame count * frame size 2635 - activeSleepTime from activeSleepTimeUs() 2636 - idleSleepTime from idleSleepTimeUs() 2637 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2638 - maxPeriod from frame count and sample rate (MIXER only) 2639 2640The parameters that affect these derived values are: 2641 - frame count 2642 - frame size 2643 - sample rate 2644 - device type: A2DP or not 2645 - device latency 2646 - format: PCM or not 2647 - active sleep time 2648 - idle sleep time 2649*/ 2650 2651void AudioFlinger::PlaybackThread::cacheParameters_l() 2652{ 2653 mixBufferSize = mFrameCount * mFrameSize; 2654 activeSleepTime = activeSleepTimeUs(); 2655 idleSleepTime = idleSleepTimeUs(); 2656} 2657 2658void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2659{ 2660 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2661 this, streamType, mTracks.size()); 2662 Mutex::Autolock _l(mLock); 2663 2664 size_t size = mTracks.size(); 2665 for (size_t i = 0; i < size; i++) { 2666 sp<Track> t = mTracks[i]; 2667 if (t->streamType() == streamType) { 2668 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2669 t->mCblk->cv.signal(); 2670 } 2671 } 2672} 2673 2674void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2675{ 2676 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2677 this, streamType, valid); 2678 Mutex::Autolock _l(mLock); 2679 2680 mStreamTypes[streamType].valid = valid; 2681} 2682 2683// getTrackName_l() must be called with ThreadBase::mLock held 2684int AudioFlinger::MixerThread::getTrackName_l() 2685{ 2686 return mAudioMixer->getTrackName(); 2687} 2688 2689// deleteTrackName_l() must be called with ThreadBase::mLock held 2690void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2691{ 2692 ALOGV("remove track (%d) and delete from mixer", name); 2693 mAudioMixer->deleteTrackName(name); 2694} 2695 2696// checkForNewParameters_l() must be called with ThreadBase::mLock held 2697bool AudioFlinger::MixerThread::checkForNewParameters_l() 2698{ 2699 bool reconfig = false; 2700 2701 while (!mNewParameters.isEmpty()) { 2702 status_t status = NO_ERROR; 2703 String8 keyValuePair = mNewParameters[0]; 2704 AudioParameter param = AudioParameter(keyValuePair); 2705 int value; 2706 2707 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2708 reconfig = true; 2709 } 2710 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2711 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2712 status = BAD_VALUE; 2713 } else { 2714 reconfig = true; 2715 } 2716 } 2717 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2718 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2719 status = BAD_VALUE; 2720 } else { 2721 reconfig = true; 2722 } 2723 } 2724 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2725 // do not accept frame count changes if tracks are open as the track buffer 2726 // size depends on frame count and correct behavior would not be guaranteed 2727 // if frame count is changed after track creation 2728 if (!mTracks.isEmpty()) { 2729 status = INVALID_OPERATION; 2730 } else { 2731 reconfig = true; 2732 } 2733 } 2734 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2735#ifdef ADD_BATTERY_DATA 2736 // when changing the audio output device, call addBatteryData to notify 2737 // the change 2738 if ((int)mDevice != value) { 2739 uint32_t params = 0; 2740 // check whether speaker is on 2741 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2742 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2743 } 2744 2745 int deviceWithoutSpeaker 2746 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2747 // check if any other device (except speaker) is on 2748 if (value & deviceWithoutSpeaker ) { 2749 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2750 } 2751 2752 if (params != 0) { 2753 addBatteryData(params); 2754 } 2755 } 2756#endif 2757 2758 // forward device change to effects that have requested to be 2759 // aware of attached audio device. 2760 mDevice = (uint32_t)value; 2761 for (size_t i = 0; i < mEffectChains.size(); i++) { 2762 mEffectChains[i]->setDevice_l(mDevice); 2763 } 2764 } 2765 2766 if (status == NO_ERROR) { 2767 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2768 keyValuePair.string()); 2769 if (!mStandby && status == INVALID_OPERATION) { 2770 mOutput->stream->common.standby(&mOutput->stream->common); 2771 mStandby = true; 2772 mBytesWritten = 0; 2773 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2774 keyValuePair.string()); 2775 } 2776 if (status == NO_ERROR && reconfig) { 2777 delete mAudioMixer; 2778 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2779 mAudioMixer = NULL; 2780 readOutputParameters(); 2781 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2782 for (size_t i = 0; i < mTracks.size() ; i++) { 2783 int name = getTrackName_l(); 2784 if (name < 0) break; 2785 mTracks[i]->mName = name; 2786 // limit track sample rate to 2 x new output sample rate 2787 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2788 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2789 } 2790 } 2791 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2792 } 2793 } 2794 2795 mNewParameters.removeAt(0); 2796 2797 mParamStatus = status; 2798 mParamCond.signal(); 2799 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2800 // already timed out waiting for the status and will never signal the condition. 2801 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2802 } 2803 return reconfig; 2804} 2805 2806status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2807{ 2808 const size_t SIZE = 256; 2809 char buffer[SIZE]; 2810 String8 result; 2811 2812 PlaybackThread::dumpInternals(fd, args); 2813 2814 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2815 result.append(buffer); 2816 write(fd, result.string(), result.size()); 2817 return NO_ERROR; 2818} 2819 2820uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2821{ 2822 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2823} 2824 2825uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2826{ 2827 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2828} 2829 2830void AudioFlinger::MixerThread::cacheParameters_l() 2831{ 2832 PlaybackThread::cacheParameters_l(); 2833 2834 // FIXME: Relaxed timing because of a certain device that can't meet latency 2835 // Should be reduced to 2x after the vendor fixes the driver issue 2836 // increase threshold again due to low power audio mode. The way this warning 2837 // threshold is calculated and its usefulness should be reconsidered anyway. 2838 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2839} 2840 2841// ---------------------------------------------------------------------------- 2842AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2843 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2844 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2845 // mLeftVolFloat, mRightVolFloat 2846 // mLeftVolShort, mRightVolShort 2847{ 2848} 2849 2850AudioFlinger::DirectOutputThread::~DirectOutputThread() 2851{ 2852} 2853 2854AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2855 Vector< sp<Track> > *tracksToRemove 2856) 2857{ 2858 sp<Track> trackToRemove; 2859 2860 mixer_state mixerStatus = MIXER_IDLE; 2861 2862 // find out which tracks need to be processed 2863 if (mActiveTracks.size() != 0) { 2864 sp<Track> t = mActiveTracks[0].promote(); 2865 // The track died recently 2866 if (t == 0) return MIXER_IDLE; 2867 2868 Track* const track = t.get(); 2869 audio_track_cblk_t* cblk = track->cblk(); 2870 2871 // The first time a track is added we wait 2872 // for all its buffers to be filled before processing it 2873 if (cblk->framesReady() && track->isReady() && 2874 !track->isPaused() && !track->isTerminated()) 2875 { 2876 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2877 2878 if (track->mFillingUpStatus == Track::FS_FILLED) { 2879 track->mFillingUpStatus = Track::FS_ACTIVE; 2880 mLeftVolFloat = mRightVolFloat = 0; 2881 mLeftVolShort = mRightVolShort = 0; 2882 if (track->mState == TrackBase::RESUMING) { 2883 track->mState = TrackBase::ACTIVE; 2884 rampVolume = true; 2885 } 2886 } else if (cblk->server != 0) { 2887 // If the track is stopped before the first frame was mixed, 2888 // do not apply ramp 2889 rampVolume = true; 2890 } 2891 // compute volume for this track 2892 float left, right; 2893 if (track->isMuted() || mMasterMute || track->isPausing() || 2894 mStreamTypes[track->streamType()].mute) { 2895 left = right = 0; 2896 if (track->isPausing()) { 2897 track->setPaused(); 2898 } 2899 } else { 2900 float typeVolume = mStreamTypes[track->streamType()].volume; 2901 float v = mMasterVolume * typeVolume; 2902 uint32_t vlr = cblk->getVolumeLR(); 2903 float v_clamped = v * (vlr & 0xFFFF); 2904 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2905 left = v_clamped/MAX_GAIN; 2906 v_clamped = v * (vlr >> 16); 2907 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2908 right = v_clamped/MAX_GAIN; 2909 } 2910 2911 if (left != mLeftVolFloat || right != mRightVolFloat) { 2912 mLeftVolFloat = left; 2913 mRightVolFloat = right; 2914 2915 // If audio HAL implements volume control, 2916 // force software volume to nominal value 2917 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2918 left = 1.0f; 2919 right = 1.0f; 2920 } 2921 2922 // Convert volumes from float to 8.24 2923 uint32_t vl = (uint32_t)(left * (1 << 24)); 2924 uint32_t vr = (uint32_t)(right * (1 << 24)); 2925 2926 // Delegate volume control to effect in track effect chain if needed 2927 // only one effect chain can be present on DirectOutputThread, so if 2928 // there is one, the track is connected to it 2929 if (!mEffectChains.isEmpty()) { 2930 // Do not ramp volume if volume is controlled by effect 2931 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2932 rampVolume = false; 2933 } 2934 } 2935 2936 // Convert volumes from 8.24 to 4.12 format 2937 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2938 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2939 leftVol = (uint16_t)v_clamped; 2940 v_clamped = (vr + (1 << 11)) >> 12; 2941 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2942 rightVol = (uint16_t)v_clamped; 2943 } else { 2944 leftVol = mLeftVolShort; 2945 rightVol = mRightVolShort; 2946 rampVolume = false; 2947 } 2948 2949 // reset retry count 2950 track->mRetryCount = kMaxTrackRetriesDirect; 2951 mActiveTrack = t; 2952 mixerStatus = MIXER_TRACKS_READY; 2953 } else { 2954 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2955 if (track->isStopped()) { 2956 track->reset(); 2957 } 2958 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2959 // We have consumed all the buffers of this track. 2960 // Remove it from the list of active tracks. 2961 // TODO: implement behavior for compressed audio 2962 size_t audioHALFrames = 2963 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2964 size_t framesWritten = 2965 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2966 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2967 trackToRemove = track; 2968 } 2969 } else { 2970 // No buffers for this track. Give it a few chances to 2971 // fill a buffer, then remove it from active list. 2972 if (--(track->mRetryCount) <= 0) { 2973 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2974 trackToRemove = track; 2975 } else { 2976 mixerStatus = MIXER_TRACKS_ENABLED; 2977 } 2978 } 2979 } 2980 } 2981 2982 // FIXME merge this with similar code for removing multiple tracks 2983 // remove all the tracks that need to be... 2984 if (CC_UNLIKELY(trackToRemove != 0)) { 2985 tracksToRemove->add(trackToRemove); 2986 mActiveTracks.remove(trackToRemove); 2987 if (!mEffectChains.isEmpty()) { 2988 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2989 trackToRemove->sessionId()); 2990 mEffectChains[0]->decActiveTrackCnt(); 2991 } 2992 if (trackToRemove->isTerminated()) { 2993 removeTrack_l(trackToRemove); 2994 } 2995 } 2996 2997 return mixerStatus; 2998} 2999 3000void AudioFlinger::DirectOutputThread::threadLoop_mix() 3001{ 3002 AudioBufferProvider::Buffer buffer; 3003 size_t frameCount = mFrameCount; 3004 int8_t *curBuf = (int8_t *)mMixBuffer; 3005 // output audio to hardware 3006 while (frameCount) { 3007 buffer.frameCount = frameCount; 3008 mActiveTrack->getNextBuffer(&buffer); 3009 if (CC_UNLIKELY(buffer.raw == NULL)) { 3010 memset(curBuf, 0, frameCount * mFrameSize); 3011 break; 3012 } 3013 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3014 frameCount -= buffer.frameCount; 3015 curBuf += buffer.frameCount * mFrameSize; 3016 mActiveTrack->releaseBuffer(&buffer); 3017 } 3018 sleepTime = 0; 3019 standbyTime = systemTime() + standbyDelay; 3020 mActiveTrack.clear(); 3021 3022 // apply volume 3023 3024 // Do not apply volume on compressed audio 3025 if (!audio_is_linear_pcm(mFormat)) { 3026 return; 3027 } 3028 3029 // convert to signed 16 bit before volume calculation 3030 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3031 size_t count = mFrameCount * mChannelCount; 3032 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3033 int16_t *dst = mMixBuffer + count-1; 3034 while (count--) { 3035 *dst-- = (int16_t)(*src--^0x80) << 8; 3036 } 3037 } 3038 3039 frameCount = mFrameCount; 3040 int16_t *out = mMixBuffer; 3041 if (rampVolume) { 3042 if (mChannelCount == 1) { 3043 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3044 int32_t vlInc = d / (int32_t)frameCount; 3045 int32_t vl = ((int32_t)mLeftVolShort << 16); 3046 do { 3047 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3048 out++; 3049 vl += vlInc; 3050 } while (--frameCount); 3051 3052 } else { 3053 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3054 int32_t vlInc = d / (int32_t)frameCount; 3055 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3056 int32_t vrInc = d / (int32_t)frameCount; 3057 int32_t vl = ((int32_t)mLeftVolShort << 16); 3058 int32_t vr = ((int32_t)mRightVolShort << 16); 3059 do { 3060 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3061 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3062 out += 2; 3063 vl += vlInc; 3064 vr += vrInc; 3065 } while (--frameCount); 3066 } 3067 } else { 3068 if (mChannelCount == 1) { 3069 do { 3070 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3071 out++; 3072 } while (--frameCount); 3073 } else { 3074 do { 3075 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3076 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3077 out += 2; 3078 } while (--frameCount); 3079 } 3080 } 3081 3082 // convert back to unsigned 8 bit after volume calculation 3083 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3084 size_t count = mFrameCount * mChannelCount; 3085 int16_t *src = mMixBuffer; 3086 uint8_t *dst = (uint8_t *)mMixBuffer; 3087 while (count--) { 3088 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3089 } 3090 } 3091 3092 mLeftVolShort = leftVol; 3093 mRightVolShort = rightVol; 3094} 3095 3096void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3097{ 3098 if (sleepTime == 0) { 3099 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3100 sleepTime = activeSleepTime; 3101 } else { 3102 sleepTime = idleSleepTime; 3103 } 3104 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3105 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3106 sleepTime = 0; 3107 } 3108} 3109 3110// getTrackName_l() must be called with ThreadBase::mLock held 3111int AudioFlinger::DirectOutputThread::getTrackName_l() 3112{ 3113 return 0; 3114} 3115 3116// deleteTrackName_l() must be called with ThreadBase::mLock held 3117void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3118{ 3119} 3120 3121// checkForNewParameters_l() must be called with ThreadBase::mLock held 3122bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3123{ 3124 bool reconfig = false; 3125 3126 while (!mNewParameters.isEmpty()) { 3127 status_t status = NO_ERROR; 3128 String8 keyValuePair = mNewParameters[0]; 3129 AudioParameter param = AudioParameter(keyValuePair); 3130 int value; 3131 3132 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3133 // do not accept frame count changes if tracks are open as the track buffer 3134 // size depends on frame count and correct behavior would not be garantied 3135 // if frame count is changed after track creation 3136 if (!mTracks.isEmpty()) { 3137 status = INVALID_OPERATION; 3138 } else { 3139 reconfig = true; 3140 } 3141 } 3142 if (status == NO_ERROR) { 3143 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3144 keyValuePair.string()); 3145 if (!mStandby && status == INVALID_OPERATION) { 3146 mOutput->stream->common.standby(&mOutput->stream->common); 3147 mStandby = true; 3148 mBytesWritten = 0; 3149 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3150 keyValuePair.string()); 3151 } 3152 if (status == NO_ERROR && reconfig) { 3153 readOutputParameters(); 3154 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3155 } 3156 } 3157 3158 mNewParameters.removeAt(0); 3159 3160 mParamStatus = status; 3161 mParamCond.signal(); 3162 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3163 // already timed out waiting for the status and will never signal the condition. 3164 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3165 } 3166 return reconfig; 3167} 3168 3169uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3170{ 3171 uint32_t time; 3172 if (audio_is_linear_pcm(mFormat)) { 3173 time = PlaybackThread::activeSleepTimeUs(); 3174 } else { 3175 time = 10000; 3176 } 3177 return time; 3178} 3179 3180uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3181{ 3182 uint32_t time; 3183 if (audio_is_linear_pcm(mFormat)) { 3184 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3185 } else { 3186 time = 10000; 3187 } 3188 return time; 3189} 3190 3191uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3192{ 3193 uint32_t time; 3194 if (audio_is_linear_pcm(mFormat)) { 3195 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3196 } else { 3197 time = 10000; 3198 } 3199 return time; 3200} 3201 3202void AudioFlinger::DirectOutputThread::cacheParameters_l() 3203{ 3204 PlaybackThread::cacheParameters_l(); 3205 3206 // use shorter standby delay as on normal output to release 3207 // hardware resources as soon as possible 3208 standbyDelay = microseconds(activeSleepTime*2); 3209} 3210 3211// ---------------------------------------------------------------------------- 3212 3213AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3214 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3215 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3216 mWaitTimeMs(UINT_MAX) 3217{ 3218 addOutputTrack(mainThread); 3219} 3220 3221AudioFlinger::DuplicatingThread::~DuplicatingThread() 3222{ 3223 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3224 mOutputTracks[i]->destroy(); 3225 } 3226} 3227 3228void AudioFlinger::DuplicatingThread::threadLoop_mix() 3229{ 3230 // mix buffers... 3231 if (outputsReady(outputTracks)) { 3232 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3233 } else { 3234 memset(mMixBuffer, 0, mixBufferSize); 3235 } 3236 sleepTime = 0; 3237 writeFrames = mFrameCount; 3238} 3239 3240void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3241{ 3242 if (sleepTime == 0) { 3243 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3244 sleepTime = activeSleepTime; 3245 } else { 3246 sleepTime = idleSleepTime; 3247 } 3248 } else if (mBytesWritten != 0) { 3249 // flush remaining overflow buffers in output tracks 3250 for (size_t i = 0; i < outputTracks.size(); i++) { 3251 if (outputTracks[i]->isActive()) { 3252 sleepTime = 0; 3253 writeFrames = 0; 3254 memset(mMixBuffer, 0, mixBufferSize); 3255 break; 3256 } 3257 } 3258 } 3259} 3260 3261void AudioFlinger::DuplicatingThread::threadLoop_write() 3262{ 3263 standbyTime = systemTime() + standbyDelay; 3264 for (size_t i = 0; i < outputTracks.size(); i++) { 3265 outputTracks[i]->write(mMixBuffer, writeFrames); 3266 } 3267 mBytesWritten += mixBufferSize; 3268} 3269 3270void AudioFlinger::DuplicatingThread::threadLoop_standby() 3271{ 3272 // DuplicatingThread implements standby by stopping all tracks 3273 for (size_t i = 0; i < outputTracks.size(); i++) { 3274 outputTracks[i]->stop(); 3275 } 3276} 3277 3278void AudioFlinger::DuplicatingThread::saveOutputTracks() 3279{ 3280 outputTracks = mOutputTracks; 3281} 3282 3283void AudioFlinger::DuplicatingThread::clearOutputTracks() 3284{ 3285 outputTracks.clear(); 3286} 3287 3288void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3289{ 3290 Mutex::Autolock _l(mLock); 3291 // FIXME explain this formula 3292 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3293 OutputTrack *outputTrack = new OutputTrack(thread, 3294 this, 3295 mSampleRate, 3296 mFormat, 3297 mChannelMask, 3298 frameCount); 3299 if (outputTrack->cblk() != NULL) { 3300 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3301 mOutputTracks.add(outputTrack); 3302 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3303 updateWaitTime_l(); 3304 } 3305} 3306 3307void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3308{ 3309 Mutex::Autolock _l(mLock); 3310 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3311 if (mOutputTracks[i]->thread() == thread) { 3312 mOutputTracks[i]->destroy(); 3313 mOutputTracks.removeAt(i); 3314 updateWaitTime_l(); 3315 return; 3316 } 3317 } 3318 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3319} 3320 3321// caller must hold mLock 3322void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3323{ 3324 mWaitTimeMs = UINT_MAX; 3325 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3326 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3327 if (strong != 0) { 3328 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3329 if (waitTimeMs < mWaitTimeMs) { 3330 mWaitTimeMs = waitTimeMs; 3331 } 3332 } 3333 } 3334} 3335 3336 3337bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3338{ 3339 for (size_t i = 0; i < outputTracks.size(); i++) { 3340 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3341 if (thread == 0) { 3342 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3343 return false; 3344 } 3345 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3346 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3347 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3348 return false; 3349 } 3350 } 3351 return true; 3352} 3353 3354uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3355{ 3356 return (mWaitTimeMs * 1000) / 2; 3357} 3358 3359void AudioFlinger::DuplicatingThread::cacheParameters_l() 3360{ 3361 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3362 updateWaitTime_l(); 3363 3364 MixerThread::cacheParameters_l(); 3365} 3366 3367// ---------------------------------------------------------------------------- 3368 3369// TrackBase constructor must be called with AudioFlinger::mLock held 3370AudioFlinger::ThreadBase::TrackBase::TrackBase( 3371 ThreadBase *thread, 3372 const sp<Client>& client, 3373 uint32_t sampleRate, 3374 audio_format_t format, 3375 uint32_t channelMask, 3376 int frameCount, 3377 const sp<IMemory>& sharedBuffer, 3378 int sessionId) 3379 : RefBase(), 3380 mThread(thread), 3381 mClient(client), 3382 mCblk(NULL), 3383 // mBuffer 3384 // mBufferEnd 3385 mFrameCount(0), 3386 mState(IDLE), 3387 mFormat(format), 3388 mStepServerFailed(false), 3389 mSessionId(sessionId) 3390 // mChannelCount 3391 // mChannelMask 3392{ 3393 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3394 3395 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3396 size_t size = sizeof(audio_track_cblk_t); 3397 uint8_t channelCount = popcount(channelMask); 3398 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3399 if (sharedBuffer == 0) { 3400 size += bufferSize; 3401 } 3402 3403 if (client != NULL) { 3404 mCblkMemory = client->heap()->allocate(size); 3405 if (mCblkMemory != 0) { 3406 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3407 if (mCblk != NULL) { // construct the shared structure in-place. 3408 new(mCblk) audio_track_cblk_t(); 3409 // clear all buffers 3410 mCblk->frameCount = frameCount; 3411 mCblk->sampleRate = sampleRate; 3412// uncomment the following lines to quickly test 32-bit wraparound 3413// mCblk->user = 0xffff0000; 3414// mCblk->server = 0xffff0000; 3415// mCblk->userBase = 0xffff0000; 3416// mCblk->serverBase = 0xffff0000; 3417 mChannelCount = channelCount; 3418 mChannelMask = channelMask; 3419 if (sharedBuffer == 0) { 3420 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3421 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3422 // Force underrun condition to avoid false underrun callback until first data is 3423 // written to buffer (other flags are cleared) 3424 mCblk->flags = CBLK_UNDERRUN_ON; 3425 } else { 3426 mBuffer = sharedBuffer->pointer(); 3427 } 3428 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3429 } 3430 } else { 3431 ALOGE("not enough memory for AudioTrack size=%u", size); 3432 client->heap()->dump("AudioTrack"); 3433 return; 3434 } 3435 } else { 3436 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3437 // construct the shared structure in-place. 3438 new(mCblk) audio_track_cblk_t(); 3439 // clear all buffers 3440 mCblk->frameCount = frameCount; 3441 mCblk->sampleRate = sampleRate; 3442// uncomment the following lines to quickly test 32-bit wraparound 3443// mCblk->user = 0xffff0000; 3444// mCblk->server = 0xffff0000; 3445// mCblk->userBase = 0xffff0000; 3446// mCblk->serverBase = 0xffff0000; 3447 mChannelCount = channelCount; 3448 mChannelMask = channelMask; 3449 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3450 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3451 // Force underrun condition to avoid false underrun callback until first data is 3452 // written to buffer (other flags are cleared) 3453 mCblk->flags = CBLK_UNDERRUN_ON; 3454 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3455 } 3456} 3457 3458AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3459{ 3460 if (mCblk != NULL) { 3461 if (mClient == 0) { 3462 delete mCblk; 3463 } else { 3464 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3465 } 3466 } 3467 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3468 if (mClient != 0) { 3469 // Client destructor must run with AudioFlinger mutex locked 3470 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3471 // If the client's reference count drops to zero, the associated destructor 3472 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3473 // relying on the automatic clear() at end of scope. 3474 mClient.clear(); 3475 } 3476} 3477 3478// AudioBufferProvider interface 3479// getNextBuffer() = 0; 3480// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3481void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3482{ 3483 buffer->raw = NULL; 3484 mFrameCount = buffer->frameCount; 3485 (void) step(); // ignore return value of step() 3486 buffer->frameCount = 0; 3487} 3488 3489bool AudioFlinger::ThreadBase::TrackBase::step() { 3490 bool result; 3491 audio_track_cblk_t* cblk = this->cblk(); 3492 3493 result = cblk->stepServer(mFrameCount); 3494 if (!result) { 3495 ALOGV("stepServer failed acquiring cblk mutex"); 3496 mStepServerFailed = true; 3497 } 3498 return result; 3499} 3500 3501void AudioFlinger::ThreadBase::TrackBase::reset() { 3502 audio_track_cblk_t* cblk = this->cblk(); 3503 3504 cblk->user = 0; 3505 cblk->server = 0; 3506 cblk->userBase = 0; 3507 cblk->serverBase = 0; 3508 mStepServerFailed = false; 3509 ALOGV("TrackBase::reset"); 3510} 3511 3512int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3513 return (int)mCblk->sampleRate; 3514} 3515 3516void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3517 audio_track_cblk_t* cblk = this->cblk(); 3518 size_t frameSize = cblk->frameSize; 3519 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3520 int8_t *bufferEnd = bufferStart + frames * frameSize; 3521 3522 // Check validity of returned pointer in case the track control block would have been corrupted. 3523 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3524 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3525 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3526 server %u, serverBase %u, user %u, userBase %u", 3527 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3528 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3529 return NULL; 3530 } 3531 3532 return bufferStart; 3533} 3534 3535status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3536{ 3537 mSyncEvents.add(event); 3538 return NO_ERROR; 3539} 3540 3541// ---------------------------------------------------------------------------- 3542 3543// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3544AudioFlinger::PlaybackThread::Track::Track( 3545 PlaybackThread *thread, 3546 const sp<Client>& client, 3547 audio_stream_type_t streamType, 3548 uint32_t sampleRate, 3549 audio_format_t format, 3550 uint32_t channelMask, 3551 int frameCount, 3552 const sp<IMemory>& sharedBuffer, 3553 int sessionId) 3554 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3555 mMute(false), 3556 // mFillingUpStatus ? 3557 // mRetryCount initialized later when needed 3558 mSharedBuffer(sharedBuffer), 3559 mStreamType(streamType), 3560 mName(-1), // see note below 3561 mMainBuffer(thread->mixBuffer()), 3562 mAuxBuffer(NULL), 3563 mAuxEffectId(0), mHasVolumeController(false), 3564 mPresentationCompleteFrames(0) 3565{ 3566 if (mCblk != NULL) { 3567 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3568 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3569 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3570 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3571 mName = thread->getTrackName_l(); 3572 if (mName < 0) { 3573 ALOGE("no more track names available"); 3574 } 3575 } 3576 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3577} 3578 3579AudioFlinger::PlaybackThread::Track::~Track() 3580{ 3581 ALOGV("PlaybackThread::Track destructor"); 3582 sp<ThreadBase> thread = mThread.promote(); 3583 if (thread != 0) { 3584 Mutex::Autolock _l(thread->mLock); 3585 mState = TERMINATED; 3586 } 3587} 3588 3589void AudioFlinger::PlaybackThread::Track::destroy() 3590{ 3591 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3592 // by removing it from mTracks vector, so there is a risk that this Tracks's 3593 // destructor is called. As the destructor needs to lock mLock, 3594 // we must acquire a strong reference on this Track before locking mLock 3595 // here so that the destructor is called only when exiting this function. 3596 // On the other hand, as long as Track::destroy() is only called by 3597 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3598 // this Track with its member mTrack. 3599 sp<Track> keep(this); 3600 { // scope for mLock 3601 sp<ThreadBase> thread = mThread.promote(); 3602 if (thread != 0) { 3603 if (!isOutputTrack()) { 3604 if (mState == ACTIVE || mState == RESUMING) { 3605 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3606 3607#ifdef ADD_BATTERY_DATA 3608 // to track the speaker usage 3609 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3610#endif 3611 } 3612 AudioSystem::releaseOutput(thread->id()); 3613 } 3614 Mutex::Autolock _l(thread->mLock); 3615 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3616 playbackThread->destroyTrack_l(this); 3617 } 3618 } 3619} 3620 3621void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3622{ 3623 uint32_t vlr = mCblk->getVolumeLR(); 3624 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", 3625 mName - AudioMixer::TRACK0, 3626 (mClient == 0) ? getpid_cached : mClient->pid(), 3627 mStreamType, 3628 mFormat, 3629 mChannelMask, 3630 mSessionId, 3631 mFrameCount, 3632 mState, 3633 mMute, 3634 mFillingUpStatus, 3635 mCblk->sampleRate, 3636 vlr & 0xFFFF, 3637 vlr >> 16, 3638 mCblk->server, 3639 mCblk->user, 3640 (int)mMainBuffer, 3641 (int)mAuxBuffer); 3642} 3643 3644// AudioBufferProvider interface 3645status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3646 AudioBufferProvider::Buffer* buffer, int64_t pts) 3647{ 3648 audio_track_cblk_t* cblk = this->cblk(); 3649 uint32_t framesReady; 3650 uint32_t framesReq = buffer->frameCount; 3651 3652 // Check if last stepServer failed, try to step now 3653 if (mStepServerFailed) { 3654 if (!step()) goto getNextBuffer_exit; 3655 ALOGV("stepServer recovered"); 3656 mStepServerFailed = false; 3657 } 3658 3659 framesReady = cblk->framesReady(); 3660 3661 if (CC_LIKELY(framesReady)) { 3662 uint32_t s = cblk->server; 3663 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3664 3665 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3666 if (framesReq > framesReady) { 3667 framesReq = framesReady; 3668 } 3669 if (framesReq > bufferEnd - s) { 3670 framesReq = bufferEnd - s; 3671 } 3672 3673 buffer->raw = getBuffer(s, framesReq); 3674 if (buffer->raw == NULL) goto getNextBuffer_exit; 3675 3676 buffer->frameCount = framesReq; 3677 return NO_ERROR; 3678 } 3679 3680getNextBuffer_exit: 3681 buffer->raw = NULL; 3682 buffer->frameCount = 0; 3683 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3684 return NOT_ENOUGH_DATA; 3685} 3686 3687uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3688 return mCblk->framesReady(); 3689} 3690 3691bool AudioFlinger::PlaybackThread::Track::isReady() const { 3692 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3693 3694 if (framesReady() >= mCblk->frameCount || 3695 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3696 mFillingUpStatus = FS_FILLED; 3697 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3698 return true; 3699 } 3700 return false; 3701} 3702 3703status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid, 3704 AudioSystem::sync_event_t event, 3705 int triggerSession) 3706{ 3707 status_t status = NO_ERROR; 3708 ALOGV("start(%d), calling pid %d session %d tid %d", 3709 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3710 sp<ThreadBase> thread = mThread.promote(); 3711 if (thread != 0) { 3712 Mutex::Autolock _l(thread->mLock); 3713 track_state state = mState; 3714 // here the track could be either new, or restarted 3715 // in both cases "unstop" the track 3716 if (mState == PAUSED) { 3717 mState = TrackBase::RESUMING; 3718 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3719 } else { 3720 mState = TrackBase::ACTIVE; 3721 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3722 } 3723 3724 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3725 thread->mLock.unlock(); 3726 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3727 thread->mLock.lock(); 3728 3729#ifdef ADD_BATTERY_DATA 3730 // to track the speaker usage 3731 if (status == NO_ERROR) { 3732 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3733 } 3734#endif 3735 } 3736 if (status == NO_ERROR) { 3737 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3738 playbackThread->addTrack_l(this); 3739 } else { 3740 mState = state; 3741 } 3742 } else { 3743 status = BAD_VALUE; 3744 } 3745 return status; 3746} 3747 3748void AudioFlinger::PlaybackThread::Track::stop() 3749{ 3750 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3751 sp<ThreadBase> thread = mThread.promote(); 3752 if (thread != 0) { 3753 Mutex::Autolock _l(thread->mLock); 3754 track_state state = mState; 3755 if (mState > STOPPED) { 3756 mState = STOPPED; 3757 // If the track is not active (PAUSED and buffers full), flush buffers 3758 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3759 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3760 reset(); 3761 } 3762 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3763 } 3764 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3765 thread->mLock.unlock(); 3766 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3767 thread->mLock.lock(); 3768 3769#ifdef ADD_BATTERY_DATA 3770 // to track the speaker usage 3771 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3772#endif 3773 } 3774 } 3775} 3776 3777void AudioFlinger::PlaybackThread::Track::pause() 3778{ 3779 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3780 sp<ThreadBase> thread = mThread.promote(); 3781 if (thread != 0) { 3782 Mutex::Autolock _l(thread->mLock); 3783 if (mState == ACTIVE || mState == RESUMING) { 3784 mState = PAUSING; 3785 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3786 if (!isOutputTrack()) { 3787 thread->mLock.unlock(); 3788 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3789 thread->mLock.lock(); 3790 3791#ifdef ADD_BATTERY_DATA 3792 // to track the speaker usage 3793 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3794#endif 3795 } 3796 } 3797 } 3798} 3799 3800void AudioFlinger::PlaybackThread::Track::flush() 3801{ 3802 ALOGV("flush(%d)", mName); 3803 sp<ThreadBase> thread = mThread.promote(); 3804 if (thread != 0) { 3805 Mutex::Autolock _l(thread->mLock); 3806 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3807 return; 3808 } 3809 // No point remaining in PAUSED state after a flush => go to 3810 // STOPPED state 3811 mState = STOPPED; 3812 3813 // do not reset the track if it is still in the process of being stopped or paused. 3814 // this will be done by prepareTracks_l() when the track is stopped. 3815 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3816 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3817 reset(); 3818 } 3819 } 3820} 3821 3822void AudioFlinger::PlaybackThread::Track::reset() 3823{ 3824 // Do not reset twice to avoid discarding data written just after a flush and before 3825 // the audioflinger thread detects the track is stopped. 3826 if (!mResetDone) { 3827 TrackBase::reset(); 3828 // Force underrun condition to avoid false underrun callback until first data is 3829 // written to buffer 3830 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3831 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3832 mFillingUpStatus = FS_FILLING; 3833 mResetDone = true; 3834 mPresentationCompleteFrames = 0; 3835 } 3836} 3837 3838void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3839{ 3840 mMute = muted; 3841} 3842 3843status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3844{ 3845 status_t status = DEAD_OBJECT; 3846 sp<ThreadBase> thread = mThread.promote(); 3847 if (thread != 0) { 3848 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3849 status = playbackThread->attachAuxEffect(this, EffectId); 3850 } 3851 return status; 3852} 3853 3854void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3855{ 3856 mAuxEffectId = EffectId; 3857 mAuxBuffer = buffer; 3858} 3859 3860bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 3861 size_t audioHalFrames) 3862{ 3863 // a track is considered presented when the total number of frames written to audio HAL 3864 // corresponds to the number of frames written when presentationComplete() is called for the 3865 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 3866 if (mPresentationCompleteFrames == 0) { 3867 mPresentationCompleteFrames = framesWritten + audioHalFrames; 3868 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 3869 mPresentationCompleteFrames, audioHalFrames); 3870 } 3871 if (framesWritten >= mPresentationCompleteFrames) { 3872 ALOGV("presentationComplete() session %d complete: framesWritten %d", 3873 mSessionId, framesWritten); 3874 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 3875 mPresentationCompleteFrames = 0; 3876 return true; 3877 } 3878 return false; 3879} 3880 3881void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 3882{ 3883 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 3884 if (mSyncEvents[i]->type() == type) { 3885 mSyncEvents[i]->trigger(); 3886 mSyncEvents.removeAt(i); 3887 i--; 3888 } 3889 } 3890} 3891 3892 3893// timed audio tracks 3894 3895sp<AudioFlinger::PlaybackThread::TimedTrack> 3896AudioFlinger::PlaybackThread::TimedTrack::create( 3897 PlaybackThread *thread, 3898 const sp<Client>& client, 3899 audio_stream_type_t streamType, 3900 uint32_t sampleRate, 3901 audio_format_t format, 3902 uint32_t channelMask, 3903 int frameCount, 3904 const sp<IMemory>& sharedBuffer, 3905 int sessionId) { 3906 if (!client->reserveTimedTrack()) 3907 return NULL; 3908 3909 return new TimedTrack( 3910 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3911 sharedBuffer, sessionId); 3912} 3913 3914AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3915 PlaybackThread *thread, 3916 const sp<Client>& client, 3917 audio_stream_type_t streamType, 3918 uint32_t sampleRate, 3919 audio_format_t format, 3920 uint32_t channelMask, 3921 int frameCount, 3922 const sp<IMemory>& sharedBuffer, 3923 int sessionId) 3924 : Track(thread, client, streamType, sampleRate, format, channelMask, 3925 frameCount, sharedBuffer, sessionId), 3926 mTimedSilenceBuffer(NULL), 3927 mTimedSilenceBufferSize(0), 3928 mTimedAudioOutputOnTime(false), 3929 mMediaTimeTransformValid(false) 3930{ 3931 LocalClock lc; 3932 mLocalTimeFreq = lc.getLocalFreq(); 3933 3934 mLocalTimeToSampleTransform.a_zero = 0; 3935 mLocalTimeToSampleTransform.b_zero = 0; 3936 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3937 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3938 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3939 &mLocalTimeToSampleTransform.a_to_b_denom); 3940} 3941 3942AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3943 mClient->releaseTimedTrack(); 3944 delete [] mTimedSilenceBuffer; 3945} 3946 3947status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3948 size_t size, sp<IMemory>* buffer) { 3949 3950 Mutex::Autolock _l(mTimedBufferQueueLock); 3951 3952 trimTimedBufferQueue_l(); 3953 3954 // lazily initialize the shared memory heap for timed buffers 3955 if (mTimedMemoryDealer == NULL) { 3956 const int kTimedBufferHeapSize = 512 << 10; 3957 3958 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3959 "AudioFlingerTimed"); 3960 if (mTimedMemoryDealer == NULL) 3961 return NO_MEMORY; 3962 } 3963 3964 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3965 if (newBuffer == NULL) { 3966 newBuffer = mTimedMemoryDealer->allocate(size); 3967 if (newBuffer == NULL) 3968 return NO_MEMORY; 3969 } 3970 3971 *buffer = newBuffer; 3972 return NO_ERROR; 3973} 3974 3975// caller must hold mTimedBufferQueueLock 3976void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3977 int64_t mediaTimeNow; 3978 { 3979 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3980 if (!mMediaTimeTransformValid) 3981 return; 3982 3983 int64_t targetTimeNow; 3984 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3985 ? mCCHelper.getCommonTime(&targetTimeNow) 3986 : mCCHelper.getLocalTime(&targetTimeNow); 3987 3988 if (OK != res) 3989 return; 3990 3991 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3992 &mediaTimeNow)) { 3993 return; 3994 } 3995 } 3996 3997 size_t trimIndex; 3998 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3999 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 4000 break; 4001 } 4002 4003 if (trimIndex) { 4004 mTimedBufferQueue.removeItemsAt(0, trimIndex); 4005 } 4006} 4007 4008status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4009 const sp<IMemory>& buffer, int64_t pts) { 4010 4011 { 4012 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4013 if (!mMediaTimeTransformValid) 4014 return INVALID_OPERATION; 4015 } 4016 4017 Mutex::Autolock _l(mTimedBufferQueueLock); 4018 4019 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4020 4021 return NO_ERROR; 4022} 4023 4024status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4025 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4026 4027 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 4028 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4029 target); 4030 4031 if (!(target == TimedAudioTrack::LOCAL_TIME || 4032 target == TimedAudioTrack::COMMON_TIME)) { 4033 return BAD_VALUE; 4034 } 4035 4036 Mutex::Autolock lock(mMediaTimeTransformLock); 4037 mMediaTimeTransform = xform; 4038 mMediaTimeTransformTarget = target; 4039 mMediaTimeTransformValid = true; 4040 4041 return NO_ERROR; 4042} 4043 4044#define min(a, b) ((a) < (b) ? (a) : (b)) 4045 4046// implementation of getNextBuffer for tracks whose buffers have timestamps 4047status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4048 AudioBufferProvider::Buffer* buffer, int64_t pts) 4049{ 4050 if (pts == AudioBufferProvider::kInvalidPTS) { 4051 buffer->raw = 0; 4052 buffer->frameCount = 0; 4053 return INVALID_OPERATION; 4054 } 4055 4056 Mutex::Autolock _l(mTimedBufferQueueLock); 4057 4058 while (true) { 4059 4060 // if we have no timed buffers, then fail 4061 if (mTimedBufferQueue.isEmpty()) { 4062 buffer->raw = 0; 4063 buffer->frameCount = 0; 4064 return NOT_ENOUGH_DATA; 4065 } 4066 4067 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4068 4069 // calculate the PTS of the head of the timed buffer queue expressed in 4070 // local time 4071 int64_t headLocalPTS; 4072 { 4073 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4074 4075 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4076 4077 if (mMediaTimeTransform.a_to_b_denom == 0) { 4078 // the transform represents a pause, so yield silence 4079 timedYieldSilence(buffer->frameCount, buffer); 4080 return NO_ERROR; 4081 } 4082 4083 int64_t transformedPTS; 4084 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4085 &transformedPTS)) { 4086 // the transform failed. this shouldn't happen, but if it does 4087 // then just drop this buffer 4088 ALOGW("timedGetNextBuffer transform failed"); 4089 buffer->raw = 0; 4090 buffer->frameCount = 0; 4091 mTimedBufferQueue.removeAt(0); 4092 return NO_ERROR; 4093 } 4094 4095 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4096 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4097 &headLocalPTS)) { 4098 buffer->raw = 0; 4099 buffer->frameCount = 0; 4100 return INVALID_OPERATION; 4101 } 4102 } else { 4103 headLocalPTS = transformedPTS; 4104 } 4105 } 4106 4107 // adjust the head buffer's PTS to reflect the portion of the head buffer 4108 // that has already been consumed 4109 int64_t effectivePTS = headLocalPTS + 4110 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4111 4112 // Calculate the delta in samples between the head of the input buffer 4113 // queue and the start of the next output buffer that will be written. 4114 // If the transformation fails because of over or underflow, it means 4115 // that the sample's position in the output stream is so far out of 4116 // whack that it should just be dropped. 4117 int64_t sampleDelta; 4118 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4119 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4120 mTimedBufferQueue.removeAt(0); 4121 continue; 4122 } 4123 if (!mLocalTimeToSampleTransform.doForwardTransform( 4124 (effectivePTS - pts) << 32, &sampleDelta)) { 4125 ALOGV("*** too late during sample rate transform: dropped buffer"); 4126 mTimedBufferQueue.removeAt(0); 4127 continue; 4128 } 4129 4130 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4131 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4132 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4133 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4134 4135 // if the delta between the ideal placement for the next input sample and 4136 // the current output position is within this threshold, then we will 4137 // concatenate the next input samples to the previous output 4138 const int64_t kSampleContinuityThreshold = 4139 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4140 4141 // if this is the first buffer of audio that we're emitting from this track 4142 // then it should be almost exactly on time. 4143 const int64_t kSampleStartupThreshold = 1LL << 32; 4144 4145 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4146 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4147 // the next input is close enough to being on time, so concatenate it 4148 // with the last output 4149 timedYieldSamples(buffer); 4150 4151 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4152 return NO_ERROR; 4153 } else if (sampleDelta > 0) { 4154 // the gap between the current output position and the proper start of 4155 // the next input sample is too big, so fill it with silence 4156 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4157 4158 timedYieldSilence(framesUntilNextInput, buffer); 4159 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4160 return NO_ERROR; 4161 } else { 4162 // the next input sample is late 4163 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4164 size_t onTimeSamplePosition = 4165 head.position() + lateFrames * mCblk->frameSize; 4166 4167 if (onTimeSamplePosition > head.buffer()->size()) { 4168 // all the remaining samples in the head are too late, so 4169 // drop it and move on 4170 ALOGV("*** too late: dropped buffer"); 4171 mTimedBufferQueue.removeAt(0); 4172 continue; 4173 } else { 4174 // skip over the late samples 4175 head.setPosition(onTimeSamplePosition); 4176 4177 // yield the available samples 4178 timedYieldSamples(buffer); 4179 4180 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4181 return NO_ERROR; 4182 } 4183 } 4184 } 4185} 4186 4187// Yield samples from the timed buffer queue head up to the given output 4188// buffer's capacity. 4189// 4190// Caller must hold mTimedBufferQueueLock 4191void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4192 AudioBufferProvider::Buffer* buffer) { 4193 4194 const TimedBuffer& head = mTimedBufferQueue[0]; 4195 4196 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4197 head.position()); 4198 4199 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4200 mCblk->frameSize); 4201 size_t framesRequested = buffer->frameCount; 4202 buffer->frameCount = min(framesLeftInHead, framesRequested); 4203 4204 mTimedAudioOutputOnTime = true; 4205} 4206 4207// Yield samples of silence up to the given output buffer's capacity 4208// 4209// Caller must hold mTimedBufferQueueLock 4210void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4211 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4212 4213 // lazily allocate a buffer filled with silence 4214 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4215 delete [] mTimedSilenceBuffer; 4216 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4217 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4218 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4219 } 4220 4221 buffer->raw = mTimedSilenceBuffer; 4222 size_t framesRequested = buffer->frameCount; 4223 buffer->frameCount = min(numFrames, framesRequested); 4224 4225 mTimedAudioOutputOnTime = false; 4226} 4227 4228// AudioBufferProvider interface 4229void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4230 AudioBufferProvider::Buffer* buffer) { 4231 4232 Mutex::Autolock _l(mTimedBufferQueueLock); 4233 4234 // If the buffer which was just released is part of the buffer at the head 4235 // of the queue, be sure to update the amt of the buffer which has been 4236 // consumed. If the buffer being returned is not part of the head of the 4237 // queue, its either because the buffer is part of the silence buffer, or 4238 // because the head of the timed queue was trimmed after the mixer called 4239 // getNextBuffer but before the mixer called releaseBuffer. 4240 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4241 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4242 4243 void* start = head.buffer()->pointer(); 4244 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4245 4246 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4247 head.setPosition(head.position() + 4248 (buffer->frameCount * mCblk->frameSize)); 4249 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4250 mTimedBufferQueue.removeAt(0); 4251 } 4252 } 4253 } 4254 4255 buffer->raw = 0; 4256 buffer->frameCount = 0; 4257} 4258 4259uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4260 Mutex::Autolock _l(mTimedBufferQueueLock); 4261 4262 uint32_t frames = 0; 4263 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4264 const TimedBuffer& tb = mTimedBufferQueue[i]; 4265 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4266 } 4267 4268 return frames; 4269} 4270 4271AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4272 : mPTS(0), mPosition(0) {} 4273 4274AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4275 const sp<IMemory>& buffer, int64_t pts) 4276 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4277 4278// ---------------------------------------------------------------------------- 4279 4280// RecordTrack constructor must be called with AudioFlinger::mLock held 4281AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4282 RecordThread *thread, 4283 const sp<Client>& client, 4284 uint32_t sampleRate, 4285 audio_format_t format, 4286 uint32_t channelMask, 4287 int frameCount, 4288 int sessionId) 4289 : TrackBase(thread, client, sampleRate, format, 4290 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4291 mOverflow(false) 4292{ 4293 if (mCblk != NULL) { 4294 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4295 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4296 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4297 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4298 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4299 } else { 4300 mCblk->frameSize = sizeof(int8_t); 4301 } 4302 } 4303} 4304 4305AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4306{ 4307 sp<ThreadBase> thread = mThread.promote(); 4308 if (thread != 0) { 4309 AudioSystem::releaseInput(thread->id()); 4310 } 4311} 4312 4313// AudioBufferProvider interface 4314status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4315{ 4316 audio_track_cblk_t* cblk = this->cblk(); 4317 uint32_t framesAvail; 4318 uint32_t framesReq = buffer->frameCount; 4319 4320 // Check if last stepServer failed, try to step now 4321 if (mStepServerFailed) { 4322 if (!step()) goto getNextBuffer_exit; 4323 ALOGV("stepServer recovered"); 4324 mStepServerFailed = false; 4325 } 4326 4327 framesAvail = cblk->framesAvailable_l(); 4328 4329 if (CC_LIKELY(framesAvail)) { 4330 uint32_t s = cblk->server; 4331 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4332 4333 if (framesReq > framesAvail) { 4334 framesReq = framesAvail; 4335 } 4336 if (framesReq > bufferEnd - s) { 4337 framesReq = bufferEnd - s; 4338 } 4339 4340 buffer->raw = getBuffer(s, framesReq); 4341 if (buffer->raw == NULL) goto getNextBuffer_exit; 4342 4343 buffer->frameCount = framesReq; 4344 return NO_ERROR; 4345 } 4346 4347getNextBuffer_exit: 4348 buffer->raw = NULL; 4349 buffer->frameCount = 0; 4350 return NOT_ENOUGH_DATA; 4351} 4352 4353status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid, 4354 AudioSystem::sync_event_t event, 4355 int triggerSession) 4356{ 4357 sp<ThreadBase> thread = mThread.promote(); 4358 if (thread != 0) { 4359 RecordThread *recordThread = (RecordThread *)thread.get(); 4360 return recordThread->start(this, tid, event, triggerSession); 4361 } else { 4362 return BAD_VALUE; 4363 } 4364} 4365 4366void AudioFlinger::RecordThread::RecordTrack::stop() 4367{ 4368 sp<ThreadBase> thread = mThread.promote(); 4369 if (thread != 0) { 4370 RecordThread *recordThread = (RecordThread *)thread.get(); 4371 recordThread->stop(this); 4372 TrackBase::reset(); 4373 // Force overrun condition to avoid false overrun callback until first data is 4374 // read from buffer 4375 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4376 } 4377} 4378 4379void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4380{ 4381 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4382 (mClient == 0) ? getpid_cached : mClient->pid(), 4383 mFormat, 4384 mChannelMask, 4385 mSessionId, 4386 mFrameCount, 4387 mState, 4388 mCblk->sampleRate, 4389 mCblk->server, 4390 mCblk->user); 4391} 4392 4393 4394// ---------------------------------------------------------------------------- 4395 4396AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4397 PlaybackThread *playbackThread, 4398 DuplicatingThread *sourceThread, 4399 uint32_t sampleRate, 4400 audio_format_t format, 4401 uint32_t channelMask, 4402 int frameCount) 4403 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4404 mActive(false), mSourceThread(sourceThread) 4405{ 4406 4407 if (mCblk != NULL) { 4408 mCblk->flags |= CBLK_DIRECTION_OUT; 4409 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4410 mOutBuffer.frameCount = 0; 4411 playbackThread->mTracks.add(this); 4412 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4413 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4414 mCblk, mBuffer, mCblk->buffers, 4415 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4416 } else { 4417 ALOGW("Error creating output track on thread %p", playbackThread); 4418 } 4419} 4420 4421AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4422{ 4423 clearBufferQueue(); 4424} 4425 4426status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid, 4427 AudioSystem::sync_event_t event, 4428 int triggerSession) 4429{ 4430 status_t status = Track::start(tid, event, triggerSession); 4431 if (status != NO_ERROR) { 4432 return status; 4433 } 4434 4435 mActive = true; 4436 mRetryCount = 127; 4437 return status; 4438} 4439 4440void AudioFlinger::PlaybackThread::OutputTrack::stop() 4441{ 4442 Track::stop(); 4443 clearBufferQueue(); 4444 mOutBuffer.frameCount = 0; 4445 mActive = false; 4446} 4447 4448bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4449{ 4450 Buffer *pInBuffer; 4451 Buffer inBuffer; 4452 uint32_t channelCount = mChannelCount; 4453 bool outputBufferFull = false; 4454 inBuffer.frameCount = frames; 4455 inBuffer.i16 = data; 4456 4457 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4458 4459 if (!mActive && frames != 0) { 4460 start(0); 4461 sp<ThreadBase> thread = mThread.promote(); 4462 if (thread != 0) { 4463 MixerThread *mixerThread = (MixerThread *)thread.get(); 4464 if (mCblk->frameCount > frames){ 4465 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4466 uint32_t startFrames = (mCblk->frameCount - frames); 4467 pInBuffer = new Buffer; 4468 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4469 pInBuffer->frameCount = startFrames; 4470 pInBuffer->i16 = pInBuffer->mBuffer; 4471 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4472 mBufferQueue.add(pInBuffer); 4473 } else { 4474 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4475 } 4476 } 4477 } 4478 } 4479 4480 while (waitTimeLeftMs) { 4481 // First write pending buffers, then new data 4482 if (mBufferQueue.size()) { 4483 pInBuffer = mBufferQueue.itemAt(0); 4484 } else { 4485 pInBuffer = &inBuffer; 4486 } 4487 4488 if (pInBuffer->frameCount == 0) { 4489 break; 4490 } 4491 4492 if (mOutBuffer.frameCount == 0) { 4493 mOutBuffer.frameCount = pInBuffer->frameCount; 4494 nsecs_t startTime = systemTime(); 4495 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4496 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4497 outputBufferFull = true; 4498 break; 4499 } 4500 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4501 if (waitTimeLeftMs >= waitTimeMs) { 4502 waitTimeLeftMs -= waitTimeMs; 4503 } else { 4504 waitTimeLeftMs = 0; 4505 } 4506 } 4507 4508 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4509 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4510 mCblk->stepUser(outFrames); 4511 pInBuffer->frameCount -= outFrames; 4512 pInBuffer->i16 += outFrames * channelCount; 4513 mOutBuffer.frameCount -= outFrames; 4514 mOutBuffer.i16 += outFrames * channelCount; 4515 4516 if (pInBuffer->frameCount == 0) { 4517 if (mBufferQueue.size()) { 4518 mBufferQueue.removeAt(0); 4519 delete [] pInBuffer->mBuffer; 4520 delete pInBuffer; 4521 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4522 } else { 4523 break; 4524 } 4525 } 4526 } 4527 4528 // If we could not write all frames, allocate a buffer and queue it for next time. 4529 if (inBuffer.frameCount) { 4530 sp<ThreadBase> thread = mThread.promote(); 4531 if (thread != 0 && !thread->standby()) { 4532 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4533 pInBuffer = new Buffer; 4534 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4535 pInBuffer->frameCount = inBuffer.frameCount; 4536 pInBuffer->i16 = pInBuffer->mBuffer; 4537 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4538 mBufferQueue.add(pInBuffer); 4539 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4540 } else { 4541 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4542 } 4543 } 4544 } 4545 4546 // Calling write() with a 0 length buffer, means that no more data will be written: 4547 // If no more buffers are pending, fill output track buffer to make sure it is started 4548 // by output mixer. 4549 if (frames == 0 && mBufferQueue.size() == 0) { 4550 if (mCblk->user < mCblk->frameCount) { 4551 frames = mCblk->frameCount - mCblk->user; 4552 pInBuffer = new Buffer; 4553 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4554 pInBuffer->frameCount = frames; 4555 pInBuffer->i16 = pInBuffer->mBuffer; 4556 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4557 mBufferQueue.add(pInBuffer); 4558 } else if (mActive) { 4559 stop(); 4560 } 4561 } 4562 4563 return outputBufferFull; 4564} 4565 4566status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4567{ 4568 int active; 4569 status_t result; 4570 audio_track_cblk_t* cblk = mCblk; 4571 uint32_t framesReq = buffer->frameCount; 4572 4573// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4574 buffer->frameCount = 0; 4575 4576 uint32_t framesAvail = cblk->framesAvailable(); 4577 4578 4579 if (framesAvail == 0) { 4580 Mutex::Autolock _l(cblk->lock); 4581 goto start_loop_here; 4582 while (framesAvail == 0) { 4583 active = mActive; 4584 if (CC_UNLIKELY(!active)) { 4585 ALOGV("Not active and NO_MORE_BUFFERS"); 4586 return NO_MORE_BUFFERS; 4587 } 4588 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4589 if (result != NO_ERROR) { 4590 return NO_MORE_BUFFERS; 4591 } 4592 // read the server count again 4593 start_loop_here: 4594 framesAvail = cblk->framesAvailable_l(); 4595 } 4596 } 4597 4598// if (framesAvail < framesReq) { 4599// return NO_MORE_BUFFERS; 4600// } 4601 4602 if (framesReq > framesAvail) { 4603 framesReq = framesAvail; 4604 } 4605 4606 uint32_t u = cblk->user; 4607 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4608 4609 if (framesReq > bufferEnd - u) { 4610 framesReq = bufferEnd - u; 4611 } 4612 4613 buffer->frameCount = framesReq; 4614 buffer->raw = (void *)cblk->buffer(u); 4615 return NO_ERROR; 4616} 4617 4618 4619void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4620{ 4621 size_t size = mBufferQueue.size(); 4622 4623 for (size_t i = 0; i < size; i++) { 4624 Buffer *pBuffer = mBufferQueue.itemAt(i); 4625 delete [] pBuffer->mBuffer; 4626 delete pBuffer; 4627 } 4628 mBufferQueue.clear(); 4629} 4630 4631// ---------------------------------------------------------------------------- 4632 4633AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4634 : RefBase(), 4635 mAudioFlinger(audioFlinger), 4636 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4637 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4638 mPid(pid), 4639 mTimedTrackCount(0) 4640{ 4641 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4642} 4643 4644// Client destructor must be called with AudioFlinger::mLock held 4645AudioFlinger::Client::~Client() 4646{ 4647 mAudioFlinger->removeClient_l(mPid); 4648} 4649 4650sp<MemoryDealer> AudioFlinger::Client::heap() const 4651{ 4652 return mMemoryDealer; 4653} 4654 4655// Reserve one of the limited slots for a timed audio track associated 4656// with this client 4657bool AudioFlinger::Client::reserveTimedTrack() 4658{ 4659 const int kMaxTimedTracksPerClient = 4; 4660 4661 Mutex::Autolock _l(mTimedTrackLock); 4662 4663 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4664 ALOGW("can not create timed track - pid %d has exceeded the limit", 4665 mPid); 4666 return false; 4667 } 4668 4669 mTimedTrackCount++; 4670 return true; 4671} 4672 4673// Release a slot for a timed audio track 4674void AudioFlinger::Client::releaseTimedTrack() 4675{ 4676 Mutex::Autolock _l(mTimedTrackLock); 4677 mTimedTrackCount--; 4678} 4679 4680// ---------------------------------------------------------------------------- 4681 4682AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4683 const sp<IAudioFlingerClient>& client, 4684 pid_t pid) 4685 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4686{ 4687} 4688 4689AudioFlinger::NotificationClient::~NotificationClient() 4690{ 4691} 4692 4693void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4694{ 4695 sp<NotificationClient> keep(this); 4696 mAudioFlinger->removeNotificationClient(mPid); 4697} 4698 4699// ---------------------------------------------------------------------------- 4700 4701AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4702 : BnAudioTrack(), 4703 mTrack(track) 4704{ 4705} 4706 4707AudioFlinger::TrackHandle::~TrackHandle() { 4708 // just stop the track on deletion, associated resources 4709 // will be freed from the main thread once all pending buffers have 4710 // been played. Unless it's not in the active track list, in which 4711 // case we free everything now... 4712 mTrack->destroy(); 4713} 4714 4715sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4716 return mTrack->getCblk(); 4717} 4718 4719status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4720 return mTrack->start(tid); 4721} 4722 4723void AudioFlinger::TrackHandle::stop() { 4724 mTrack->stop(); 4725} 4726 4727void AudioFlinger::TrackHandle::flush() { 4728 mTrack->flush(); 4729} 4730 4731void AudioFlinger::TrackHandle::mute(bool e) { 4732 mTrack->mute(e); 4733} 4734 4735void AudioFlinger::TrackHandle::pause() { 4736 mTrack->pause(); 4737} 4738 4739status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4740{ 4741 return mTrack->attachAuxEffect(EffectId); 4742} 4743 4744status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4745 sp<IMemory>* buffer) { 4746 if (!mTrack->isTimedTrack()) 4747 return INVALID_OPERATION; 4748 4749 PlaybackThread::TimedTrack* tt = 4750 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4751 return tt->allocateTimedBuffer(size, buffer); 4752} 4753 4754status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4755 int64_t pts) { 4756 if (!mTrack->isTimedTrack()) 4757 return INVALID_OPERATION; 4758 4759 PlaybackThread::TimedTrack* tt = 4760 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4761 return tt->queueTimedBuffer(buffer, pts); 4762} 4763 4764status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4765 const LinearTransform& xform, int target) { 4766 4767 if (!mTrack->isTimedTrack()) 4768 return INVALID_OPERATION; 4769 4770 PlaybackThread::TimedTrack* tt = 4771 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4772 return tt->setMediaTimeTransform( 4773 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4774} 4775 4776status_t AudioFlinger::TrackHandle::onTransact( 4777 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4778{ 4779 return BnAudioTrack::onTransact(code, data, reply, flags); 4780} 4781 4782// ---------------------------------------------------------------------------- 4783 4784sp<IAudioRecord> AudioFlinger::openRecord( 4785 pid_t pid, 4786 audio_io_handle_t input, 4787 uint32_t sampleRate, 4788 audio_format_t format, 4789 uint32_t channelMask, 4790 int frameCount, 4791 IAudioFlinger::track_flags_t flags, 4792 int *sessionId, 4793 status_t *status) 4794{ 4795 sp<RecordThread::RecordTrack> recordTrack; 4796 sp<RecordHandle> recordHandle; 4797 sp<Client> client; 4798 status_t lStatus; 4799 RecordThread *thread; 4800 size_t inFrameCount; 4801 int lSessionId; 4802 4803 // check calling permissions 4804 if (!recordingAllowed()) { 4805 lStatus = PERMISSION_DENIED; 4806 goto Exit; 4807 } 4808 4809 // add client to list 4810 { // scope for mLock 4811 Mutex::Autolock _l(mLock); 4812 thread = checkRecordThread_l(input); 4813 if (thread == NULL) { 4814 lStatus = BAD_VALUE; 4815 goto Exit; 4816 } 4817 4818 client = registerPid_l(pid); 4819 4820 // If no audio session id is provided, create one here 4821 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4822 lSessionId = *sessionId; 4823 } else { 4824 lSessionId = nextUniqueId(); 4825 if (sessionId != NULL) { 4826 *sessionId = lSessionId; 4827 } 4828 } 4829 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4830 recordTrack = thread->createRecordTrack_l(client, 4831 sampleRate, 4832 format, 4833 channelMask, 4834 frameCount, 4835 lSessionId, 4836 &lStatus); 4837 } 4838 if (lStatus != NO_ERROR) { 4839 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4840 // destructor is called by the TrackBase destructor with mLock held 4841 client.clear(); 4842 recordTrack.clear(); 4843 goto Exit; 4844 } 4845 4846 // return to handle to client 4847 recordHandle = new RecordHandle(recordTrack); 4848 lStatus = NO_ERROR; 4849 4850Exit: 4851 if (status) { 4852 *status = lStatus; 4853 } 4854 return recordHandle; 4855} 4856 4857// ---------------------------------------------------------------------------- 4858 4859AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4860 : BnAudioRecord(), 4861 mRecordTrack(recordTrack) 4862{ 4863} 4864 4865AudioFlinger::RecordHandle::~RecordHandle() { 4866 stop(); 4867} 4868 4869sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4870 return mRecordTrack->getCblk(); 4871} 4872 4873status_t AudioFlinger::RecordHandle::start(pid_t tid, int event, int triggerSession) { 4874 ALOGV("RecordHandle::start()"); 4875 return mRecordTrack->start(tid, (AudioSystem::sync_event_t)event, triggerSession); 4876} 4877 4878void AudioFlinger::RecordHandle::stop() { 4879 ALOGV("RecordHandle::stop()"); 4880 mRecordTrack->stop(); 4881} 4882 4883status_t AudioFlinger::RecordHandle::onTransact( 4884 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4885{ 4886 return BnAudioRecord::onTransact(code, data, reply, flags); 4887} 4888 4889// ---------------------------------------------------------------------------- 4890 4891AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4892 AudioStreamIn *input, 4893 uint32_t sampleRate, 4894 uint32_t channels, 4895 audio_io_handle_t id, 4896 uint32_t device) : 4897 ThreadBase(audioFlinger, id, device, RECORD), 4898 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4899 // mRsmpInIndex and mInputBytes set by readInputParameters() 4900 mReqChannelCount(popcount(channels)), 4901 mReqSampleRate(sampleRate) 4902 // mBytesRead is only meaningful while active, and so is cleared in start() 4903 // (but might be better to also clear here for dump?) 4904{ 4905 snprintf(mName, kNameLength, "AudioIn_%X", id); 4906 4907 readInputParameters(); 4908} 4909 4910 4911AudioFlinger::RecordThread::~RecordThread() 4912{ 4913 delete[] mRsmpInBuffer; 4914 delete mResampler; 4915 delete[] mRsmpOutBuffer; 4916} 4917 4918void AudioFlinger::RecordThread::onFirstRef() 4919{ 4920 run(mName, PRIORITY_URGENT_AUDIO); 4921} 4922 4923status_t AudioFlinger::RecordThread::readyToRun() 4924{ 4925 status_t status = initCheck(); 4926 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4927 return status; 4928} 4929 4930bool AudioFlinger::RecordThread::threadLoop() 4931{ 4932 AudioBufferProvider::Buffer buffer; 4933 sp<RecordTrack> activeTrack; 4934 Vector< sp<EffectChain> > effectChains; 4935 4936 nsecs_t lastWarning = 0; 4937 4938 acquireWakeLock(); 4939 4940 // start recording 4941 while (!exitPending()) { 4942 4943 processConfigEvents(); 4944 4945 { // scope for mLock 4946 Mutex::Autolock _l(mLock); 4947 checkForNewParameters_l(); 4948 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4949 if (!mStandby) { 4950 mInput->stream->common.standby(&mInput->stream->common); 4951 mStandby = true; 4952 } 4953 4954 if (exitPending()) break; 4955 4956 releaseWakeLock_l(); 4957 ALOGV("RecordThread: loop stopping"); 4958 // go to sleep 4959 mWaitWorkCV.wait(mLock); 4960 ALOGV("RecordThread: loop starting"); 4961 acquireWakeLock_l(); 4962 continue; 4963 } 4964 if (mActiveTrack != 0) { 4965 if (mActiveTrack->mState == TrackBase::PAUSING) { 4966 if (!mStandby) { 4967 mInput->stream->common.standby(&mInput->stream->common); 4968 mStandby = true; 4969 } 4970 mActiveTrack.clear(); 4971 mStartStopCond.broadcast(); 4972 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4973 if (mReqChannelCount != mActiveTrack->channelCount()) { 4974 mActiveTrack.clear(); 4975 mStartStopCond.broadcast(); 4976 } else if (mBytesRead != 0) { 4977 // record start succeeds only if first read from audio input 4978 // succeeds 4979 if (mBytesRead > 0) { 4980 mActiveTrack->mState = TrackBase::ACTIVE; 4981 } else { 4982 mActiveTrack.clear(); 4983 } 4984 mStartStopCond.broadcast(); 4985 } 4986 mStandby = false; 4987 } 4988 } 4989 lockEffectChains_l(effectChains); 4990 } 4991 4992 if (mActiveTrack != 0) { 4993 if (mActiveTrack->mState != TrackBase::ACTIVE && 4994 mActiveTrack->mState != TrackBase::RESUMING) { 4995 unlockEffectChains(effectChains); 4996 usleep(kRecordThreadSleepUs); 4997 continue; 4998 } 4999 for (size_t i = 0; i < effectChains.size(); i ++) { 5000 effectChains[i]->process_l(); 5001 } 5002 5003 buffer.frameCount = mFrameCount; 5004 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5005 size_t framesOut = buffer.frameCount; 5006 if (mResampler == NULL) { 5007 // no resampling 5008 while (framesOut) { 5009 size_t framesIn = mFrameCount - mRsmpInIndex; 5010 if (framesIn) { 5011 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5012 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5013 if (framesIn > framesOut) 5014 framesIn = framesOut; 5015 mRsmpInIndex += framesIn; 5016 framesOut -= framesIn; 5017 if ((int)mChannelCount == mReqChannelCount || 5018 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5019 memcpy(dst, src, framesIn * mFrameSize); 5020 } else { 5021 int16_t *src16 = (int16_t *)src; 5022 int16_t *dst16 = (int16_t *)dst; 5023 if (mChannelCount == 1) { 5024 while (framesIn--) { 5025 *dst16++ = *src16; 5026 *dst16++ = *src16++; 5027 } 5028 } else { 5029 while (framesIn--) { 5030 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5031 src16 += 2; 5032 } 5033 } 5034 } 5035 } 5036 if (framesOut && mFrameCount == mRsmpInIndex) { 5037 if (framesOut == mFrameCount && 5038 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5039 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5040 framesOut = 0; 5041 } else { 5042 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5043 mRsmpInIndex = 0; 5044 } 5045 if (mBytesRead < 0) { 5046 ALOGE("Error reading audio input"); 5047 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5048 // Force input into standby so that it tries to 5049 // recover at next read attempt 5050 mInput->stream->common.standby(&mInput->stream->common); 5051 usleep(kRecordThreadSleepUs); 5052 } 5053 mRsmpInIndex = mFrameCount; 5054 framesOut = 0; 5055 buffer.frameCount = 0; 5056 } 5057 } 5058 } 5059 } else { 5060 // resampling 5061 5062 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5063 // alter output frame count as if we were expecting stereo samples 5064 if (mChannelCount == 1 && mReqChannelCount == 1) { 5065 framesOut >>= 1; 5066 } 5067 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5068 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5069 // are 32 bit aligned which should be always true. 5070 if (mChannelCount == 2 && mReqChannelCount == 1) { 5071 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5072 // the resampler always outputs stereo samples: do post stereo to mono conversion 5073 int16_t *src = (int16_t *)mRsmpOutBuffer; 5074 int16_t *dst = buffer.i16; 5075 while (framesOut--) { 5076 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5077 src += 2; 5078 } 5079 } else { 5080 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5081 } 5082 5083 } 5084 if (mFramestoDrop == 0) { 5085 mActiveTrack->releaseBuffer(&buffer); 5086 } else { 5087 if (mFramestoDrop > 0) { 5088 mFramestoDrop -= buffer.frameCount; 5089 if (mFramestoDrop < 0) { 5090 mFramestoDrop = 0; 5091 } 5092 } 5093 } 5094 mActiveTrack->overflow(); 5095 } 5096 // client isn't retrieving buffers fast enough 5097 else { 5098 if (!mActiveTrack->setOverflow()) { 5099 nsecs_t now = systemTime(); 5100 if ((now - lastWarning) > kWarningThrottleNs) { 5101 ALOGW("RecordThread: buffer overflow"); 5102 lastWarning = now; 5103 } 5104 } 5105 // Release the processor for a while before asking for a new buffer. 5106 // This will give the application more chance to read from the buffer and 5107 // clear the overflow. 5108 usleep(kRecordThreadSleepUs); 5109 } 5110 } 5111 // enable changes in effect chain 5112 unlockEffectChains(effectChains); 5113 effectChains.clear(); 5114 } 5115 5116 if (!mStandby) { 5117 mInput->stream->common.standby(&mInput->stream->common); 5118 } 5119 mActiveTrack.clear(); 5120 5121 mStartStopCond.broadcast(); 5122 5123 releaseWakeLock(); 5124 5125 ALOGV("RecordThread %p exiting", this); 5126 return false; 5127} 5128 5129 5130sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5131 const sp<AudioFlinger::Client>& client, 5132 uint32_t sampleRate, 5133 audio_format_t format, 5134 int channelMask, 5135 int frameCount, 5136 int sessionId, 5137 status_t *status) 5138{ 5139 sp<RecordTrack> track; 5140 status_t lStatus; 5141 5142 lStatus = initCheck(); 5143 if (lStatus != NO_ERROR) { 5144 ALOGE("Audio driver not initialized."); 5145 goto Exit; 5146 } 5147 5148 { // scope for mLock 5149 Mutex::Autolock _l(mLock); 5150 5151 track = new RecordTrack(this, client, sampleRate, 5152 format, channelMask, frameCount, sessionId); 5153 5154 if (track->getCblk() == 0) { 5155 lStatus = NO_MEMORY; 5156 goto Exit; 5157 } 5158 5159 mTrack = track.get(); 5160 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5161 bool suspend = audio_is_bluetooth_sco_device( 5162 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5163 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5164 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5165 } 5166 lStatus = NO_ERROR; 5167 5168Exit: 5169 if (status) { 5170 *status = lStatus; 5171 } 5172 return track; 5173} 5174 5175status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5176 pid_t tid, AudioSystem::sync_event_t event, 5177 int triggerSession) 5178{ 5179 ALOGV("RecordThread::start tid=%d, event %d, triggerSession %d", tid, event, triggerSession); 5180 sp<ThreadBase> strongMe = this; 5181 status_t status = NO_ERROR; 5182 5183 if (event == AudioSystem::SYNC_EVENT_NONE) { 5184 mSyncStartEvent.clear(); 5185 mFramestoDrop = 0; 5186 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5187 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5188 triggerSession, 5189 recordTrack->sessionId(), 5190 syncStartEventCallback, 5191 this); 5192 mFramestoDrop = -1; 5193 } 5194 5195 { 5196 AutoMutex lock(mLock); 5197 if (mActiveTrack != 0) { 5198 if (recordTrack != mActiveTrack.get()) { 5199 status = -EBUSY; 5200 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5201 mActiveTrack->mState = TrackBase::ACTIVE; 5202 } 5203 return status; 5204 } 5205 5206 recordTrack->mState = TrackBase::IDLE; 5207 mActiveTrack = recordTrack; 5208 mLock.unlock(); 5209 status_t status = AudioSystem::startInput(mId); 5210 mLock.lock(); 5211 if (status != NO_ERROR) { 5212 mActiveTrack.clear(); 5213 clearSyncStartEvent(); 5214 return status; 5215 } 5216 mRsmpInIndex = mFrameCount; 5217 mBytesRead = 0; 5218 if (mResampler != NULL) { 5219 mResampler->reset(); 5220 } 5221 mActiveTrack->mState = TrackBase::RESUMING; 5222 // signal thread to start 5223 ALOGV("Signal record thread"); 5224 mWaitWorkCV.signal(); 5225 // do not wait for mStartStopCond if exiting 5226 if (exitPending()) { 5227 mActiveTrack.clear(); 5228 status = INVALID_OPERATION; 5229 goto startError; 5230 } 5231 mStartStopCond.wait(mLock); 5232 if (mActiveTrack == 0) { 5233 ALOGV("Record failed to start"); 5234 status = BAD_VALUE; 5235 goto startError; 5236 } 5237 ALOGV("Record started OK"); 5238 return status; 5239 } 5240startError: 5241 AudioSystem::stopInput(mId); 5242 clearSyncStartEvent(); 5243 return status; 5244} 5245 5246void AudioFlinger::RecordThread::clearSyncStartEvent() 5247{ 5248 if (mSyncStartEvent != 0) { 5249 mSyncStartEvent->cancel(); 5250 } 5251 mSyncStartEvent.clear(); 5252} 5253 5254void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5255{ 5256 sp<SyncEvent> strongEvent = event.promote(); 5257 5258 if (strongEvent != 0) { 5259 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5260 me->handleSyncStartEvent(strongEvent); 5261 } 5262} 5263 5264void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5265{ 5266 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5267 mActiveTrack.get(), 5268 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5269 event->listenerSession()); 5270 5271 if (mActiveTrack != 0 && 5272 event == mSyncStartEvent) { 5273 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5274 // from audio HAL 5275 mFramestoDrop = mFrameCount * 2; 5276 mSyncStartEvent.clear(); 5277 } 5278} 5279 5280void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5281 ALOGV("RecordThread::stop"); 5282 sp<ThreadBase> strongMe = this; 5283 { 5284 AutoMutex lock(mLock); 5285 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5286 mActiveTrack->mState = TrackBase::PAUSING; 5287 // do not wait for mStartStopCond if exiting 5288 if (exitPending()) { 5289 return; 5290 } 5291 mStartStopCond.wait(mLock); 5292 // if we have been restarted, recordTrack == mActiveTrack.get() here 5293 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5294 mLock.unlock(); 5295 AudioSystem::stopInput(mId); 5296 mLock.lock(); 5297 ALOGV("Record stopped OK"); 5298 } 5299 } 5300 } 5301} 5302 5303bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5304{ 5305 return false; 5306} 5307 5308status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5309{ 5310 if (!isValidSyncEvent(event)) { 5311 return BAD_VALUE; 5312 } 5313 5314 Mutex::Autolock _l(mLock); 5315 5316 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5317 mTrack->setSyncEvent(event); 5318 return NO_ERROR; 5319 } 5320 return NAME_NOT_FOUND; 5321} 5322 5323status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5324{ 5325 const size_t SIZE = 256; 5326 char buffer[SIZE]; 5327 String8 result; 5328 5329 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5330 result.append(buffer); 5331 5332 if (mActiveTrack != 0) { 5333 result.append("Active Track:\n"); 5334 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5335 mActiveTrack->dump(buffer, SIZE); 5336 result.append(buffer); 5337 5338 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5339 result.append(buffer); 5340 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5341 result.append(buffer); 5342 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5343 result.append(buffer); 5344 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5345 result.append(buffer); 5346 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5347 result.append(buffer); 5348 5349 5350 } else { 5351 result.append("No record client\n"); 5352 } 5353 write(fd, result.string(), result.size()); 5354 5355 dumpBase(fd, args); 5356 dumpEffectChains(fd, args); 5357 5358 return NO_ERROR; 5359} 5360 5361// AudioBufferProvider interface 5362status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5363{ 5364 size_t framesReq = buffer->frameCount; 5365 size_t framesReady = mFrameCount - mRsmpInIndex; 5366 int channelCount; 5367 5368 if (framesReady == 0) { 5369 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5370 if (mBytesRead < 0) { 5371 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5372 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5373 // Force input into standby so that it tries to 5374 // recover at next read attempt 5375 mInput->stream->common.standby(&mInput->stream->common); 5376 usleep(kRecordThreadSleepUs); 5377 } 5378 buffer->raw = NULL; 5379 buffer->frameCount = 0; 5380 return NOT_ENOUGH_DATA; 5381 } 5382 mRsmpInIndex = 0; 5383 framesReady = mFrameCount; 5384 } 5385 5386 if (framesReq > framesReady) { 5387 framesReq = framesReady; 5388 } 5389 5390 if (mChannelCount == 1 && mReqChannelCount == 2) { 5391 channelCount = 1; 5392 } else { 5393 channelCount = 2; 5394 } 5395 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5396 buffer->frameCount = framesReq; 5397 return NO_ERROR; 5398} 5399 5400// AudioBufferProvider interface 5401void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5402{ 5403 mRsmpInIndex += buffer->frameCount; 5404 buffer->frameCount = 0; 5405} 5406 5407bool AudioFlinger::RecordThread::checkForNewParameters_l() 5408{ 5409 bool reconfig = false; 5410 5411 while (!mNewParameters.isEmpty()) { 5412 status_t status = NO_ERROR; 5413 String8 keyValuePair = mNewParameters[0]; 5414 AudioParameter param = AudioParameter(keyValuePair); 5415 int value; 5416 audio_format_t reqFormat = mFormat; 5417 int reqSamplingRate = mReqSampleRate; 5418 int reqChannelCount = mReqChannelCount; 5419 5420 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5421 reqSamplingRate = value; 5422 reconfig = true; 5423 } 5424 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5425 reqFormat = (audio_format_t) value; 5426 reconfig = true; 5427 } 5428 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5429 reqChannelCount = popcount(value); 5430 reconfig = true; 5431 } 5432 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5433 // do not accept frame count changes if tracks are open as the track buffer 5434 // size depends on frame count and correct behavior would not be guaranteed 5435 // if frame count is changed after track creation 5436 if (mActiveTrack != 0) { 5437 status = INVALID_OPERATION; 5438 } else { 5439 reconfig = true; 5440 } 5441 } 5442 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5443 // forward device change to effects that have requested to be 5444 // aware of attached audio device. 5445 for (size_t i = 0; i < mEffectChains.size(); i++) { 5446 mEffectChains[i]->setDevice_l(value); 5447 } 5448 // store input device and output device but do not forward output device to audio HAL. 5449 // Note that status is ignored by the caller for output device 5450 // (see AudioFlinger::setParameters() 5451 if (value & AUDIO_DEVICE_OUT_ALL) { 5452 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5453 status = BAD_VALUE; 5454 } else { 5455 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5456 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5457 if (mTrack != NULL) { 5458 bool suspend = audio_is_bluetooth_sco_device( 5459 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5460 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5461 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5462 } 5463 } 5464 mDevice |= (uint32_t)value; 5465 } 5466 if (status == NO_ERROR) { 5467 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5468 if (status == INVALID_OPERATION) { 5469 mInput->stream->common.standby(&mInput->stream->common); 5470 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5471 keyValuePair.string()); 5472 } 5473 if (reconfig) { 5474 if (status == BAD_VALUE && 5475 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5476 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5477 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5478 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5479 (reqChannelCount <= FCC_2)) { 5480 status = NO_ERROR; 5481 } 5482 if (status == NO_ERROR) { 5483 readInputParameters(); 5484 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5485 } 5486 } 5487 } 5488 5489 mNewParameters.removeAt(0); 5490 5491 mParamStatus = status; 5492 mParamCond.signal(); 5493 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5494 // already timed out waiting for the status and will never signal the condition. 5495 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5496 } 5497 return reconfig; 5498} 5499 5500String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5501{ 5502 char *s; 5503 String8 out_s8 = String8(); 5504 5505 Mutex::Autolock _l(mLock); 5506 if (initCheck() != NO_ERROR) { 5507 return out_s8; 5508 } 5509 5510 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5511 out_s8 = String8(s); 5512 free(s); 5513 return out_s8; 5514} 5515 5516void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5517 AudioSystem::OutputDescriptor desc; 5518 void *param2 = NULL; 5519 5520 switch (event) { 5521 case AudioSystem::INPUT_OPENED: 5522 case AudioSystem::INPUT_CONFIG_CHANGED: 5523 desc.channels = mChannelMask; 5524 desc.samplingRate = mSampleRate; 5525 desc.format = mFormat; 5526 desc.frameCount = mFrameCount; 5527 desc.latency = 0; 5528 param2 = &desc; 5529 break; 5530 5531 case AudioSystem::INPUT_CLOSED: 5532 default: 5533 break; 5534 } 5535 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5536} 5537 5538void AudioFlinger::RecordThread::readInputParameters() 5539{ 5540 delete mRsmpInBuffer; 5541 // mRsmpInBuffer is always assigned a new[] below 5542 delete mRsmpOutBuffer; 5543 mRsmpOutBuffer = NULL; 5544 delete mResampler; 5545 mResampler = NULL; 5546 5547 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5548 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5549 mChannelCount = (uint16_t)popcount(mChannelMask); 5550 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5551 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5552 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5553 mFrameCount = mInputBytes / mFrameSize; 5554 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5555 5556 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5557 { 5558 int channelCount; 5559 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5560 // stereo to mono post process as the resampler always outputs stereo. 5561 if (mChannelCount == 1 && mReqChannelCount == 2) { 5562 channelCount = 1; 5563 } else { 5564 channelCount = 2; 5565 } 5566 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5567 mResampler->setSampleRate(mSampleRate); 5568 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5569 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5570 5571 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5572 if (mChannelCount == 1 && mReqChannelCount == 1) { 5573 mFrameCount >>= 1; 5574 } 5575 5576 } 5577 mRsmpInIndex = mFrameCount; 5578} 5579 5580unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5581{ 5582 Mutex::Autolock _l(mLock); 5583 if (initCheck() != NO_ERROR) { 5584 return 0; 5585 } 5586 5587 return mInput->stream->get_input_frames_lost(mInput->stream); 5588} 5589 5590uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5591{ 5592 Mutex::Autolock _l(mLock); 5593 uint32_t result = 0; 5594 if (getEffectChain_l(sessionId) != 0) { 5595 result = EFFECT_SESSION; 5596 } 5597 5598 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5599 result |= TRACK_SESSION; 5600 } 5601 5602 return result; 5603} 5604 5605AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5606{ 5607 Mutex::Autolock _l(mLock); 5608 return mTrack; 5609} 5610 5611AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5612{ 5613 Mutex::Autolock _l(mLock); 5614 return mInput; 5615} 5616 5617AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5618{ 5619 Mutex::Autolock _l(mLock); 5620 AudioStreamIn *input = mInput; 5621 mInput = NULL; 5622 return input; 5623} 5624 5625// this method must always be called either with ThreadBase mLock held or inside the thread loop 5626audio_stream_t* AudioFlinger::RecordThread::stream() 5627{ 5628 if (mInput == NULL) { 5629 return NULL; 5630 } 5631 return &mInput->stream->common; 5632} 5633 5634 5635// ---------------------------------------------------------------------------- 5636 5637audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5638 uint32_t *pSamplingRate, 5639 audio_format_t *pFormat, 5640 uint32_t *pChannels, 5641 uint32_t *pLatencyMs, 5642 audio_policy_output_flags_t flags) 5643{ 5644 status_t status; 5645 PlaybackThread *thread = NULL; 5646 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5647 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5648 uint32_t channels = pChannels ? *pChannels : 0; 5649 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5650 audio_stream_out_t *outStream; 5651 audio_hw_device_t *outHwDev; 5652 5653 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5654 pDevices ? *pDevices : 0, 5655 samplingRate, 5656 format, 5657 channels, 5658 flags); 5659 5660 if (pDevices == NULL || *pDevices == 0) { 5661 return 0; 5662 } 5663 5664 Mutex::Autolock _l(mLock); 5665 5666 outHwDev = findSuitableHwDev_l(*pDevices); 5667 if (outHwDev == NULL) 5668 return 0; 5669 5670 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5671 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5672 &channels, &samplingRate, &outStream); 5673 mHardwareStatus = AUDIO_HW_IDLE; 5674 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5675 outStream, 5676 samplingRate, 5677 format, 5678 channels, 5679 status); 5680 5681 if (outStream != NULL) { 5682 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5683 audio_io_handle_t id = nextUniqueId(); 5684 5685 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5686 (format != AUDIO_FORMAT_PCM_16_BIT) || 5687 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5688 thread = new DirectOutputThread(this, output, id, *pDevices); 5689 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5690 } else { 5691 thread = new MixerThread(this, output, id, *pDevices); 5692 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5693 } 5694 mPlaybackThreads.add(id, thread); 5695 5696 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5697 if (pFormat != NULL) *pFormat = format; 5698 if (pChannels != NULL) *pChannels = channels; 5699 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5700 5701 // notify client processes of the new output creation 5702 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5703 return id; 5704 } 5705 5706 return 0; 5707} 5708 5709audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5710 audio_io_handle_t output2) 5711{ 5712 Mutex::Autolock _l(mLock); 5713 MixerThread *thread1 = checkMixerThread_l(output1); 5714 MixerThread *thread2 = checkMixerThread_l(output2); 5715 5716 if (thread1 == NULL || thread2 == NULL) { 5717 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5718 return 0; 5719 } 5720 5721 audio_io_handle_t id = nextUniqueId(); 5722 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5723 thread->addOutputTrack(thread2); 5724 mPlaybackThreads.add(id, thread); 5725 // notify client processes of the new output creation 5726 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5727 return id; 5728} 5729 5730status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5731{ 5732 // keep strong reference on the playback thread so that 5733 // it is not destroyed while exit() is executed 5734 sp<PlaybackThread> thread; 5735 { 5736 Mutex::Autolock _l(mLock); 5737 thread = checkPlaybackThread_l(output); 5738 if (thread == NULL) { 5739 return BAD_VALUE; 5740 } 5741 5742 ALOGV("closeOutput() %d", output); 5743 5744 if (thread->type() == ThreadBase::MIXER) { 5745 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5746 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5747 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5748 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5749 } 5750 } 5751 } 5752 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5753 mPlaybackThreads.removeItem(output); 5754 } 5755 thread->exit(); 5756 // The thread entity (active unit of execution) is no longer running here, 5757 // but the ThreadBase container still exists. 5758 5759 if (thread->type() != ThreadBase::DUPLICATING) { 5760 AudioStreamOut *out = thread->clearOutput(); 5761 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5762 // from now on thread->mOutput is NULL 5763 out->hwDev->close_output_stream(out->hwDev, out->stream); 5764 delete out; 5765 } 5766 return NO_ERROR; 5767} 5768 5769status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5770{ 5771 Mutex::Autolock _l(mLock); 5772 PlaybackThread *thread = checkPlaybackThread_l(output); 5773 5774 if (thread == NULL) { 5775 return BAD_VALUE; 5776 } 5777 5778 ALOGV("suspendOutput() %d", output); 5779 thread->suspend(); 5780 5781 return NO_ERROR; 5782} 5783 5784status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5785{ 5786 Mutex::Autolock _l(mLock); 5787 PlaybackThread *thread = checkPlaybackThread_l(output); 5788 5789 if (thread == NULL) { 5790 return BAD_VALUE; 5791 } 5792 5793 ALOGV("restoreOutput() %d", output); 5794 5795 thread->restore(); 5796 5797 return NO_ERROR; 5798} 5799 5800audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5801 uint32_t *pSamplingRate, 5802 audio_format_t *pFormat, 5803 uint32_t *pChannels, 5804 audio_in_acoustics_t acoustics) 5805{ 5806 status_t status; 5807 RecordThread *thread = NULL; 5808 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5809 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5810 uint32_t channels = pChannels ? *pChannels : 0; 5811 uint32_t reqSamplingRate = samplingRate; 5812 audio_format_t reqFormat = format; 5813 uint32_t reqChannels = channels; 5814 audio_stream_in_t *inStream; 5815 audio_hw_device_t *inHwDev; 5816 5817 if (pDevices == NULL || *pDevices == 0) { 5818 return 0; 5819 } 5820 5821 Mutex::Autolock _l(mLock); 5822 5823 inHwDev = findSuitableHwDev_l(*pDevices); 5824 if (inHwDev == NULL) 5825 return 0; 5826 5827 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5828 &channels, &samplingRate, 5829 acoustics, 5830 &inStream); 5831 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5832 inStream, 5833 samplingRate, 5834 format, 5835 channels, 5836 acoustics, 5837 status); 5838 5839 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5840 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5841 // or stereo to mono conversions on 16 bit PCM inputs. 5842 if (inStream == NULL && status == BAD_VALUE && 5843 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5844 (samplingRate <= 2 * reqSamplingRate) && 5845 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5846 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5847 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5848 &channels, &samplingRate, 5849 acoustics, 5850 &inStream); 5851 } 5852 5853 if (inStream != NULL) { 5854 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5855 5856 audio_io_handle_t id = nextUniqueId(); 5857 // Start record thread 5858 // RecorThread require both input and output device indication to forward to audio 5859 // pre processing modules 5860 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5861 thread = new RecordThread(this, 5862 input, 5863 reqSamplingRate, 5864 reqChannels, 5865 id, 5866 device); 5867 mRecordThreads.add(id, thread); 5868 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5869 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5870 if (pFormat != NULL) *pFormat = format; 5871 if (pChannels != NULL) *pChannels = reqChannels; 5872 5873 input->stream->common.standby(&input->stream->common); 5874 5875 // notify client processes of the new input creation 5876 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5877 return id; 5878 } 5879 5880 return 0; 5881} 5882 5883status_t AudioFlinger::closeInput(audio_io_handle_t input) 5884{ 5885 // keep strong reference on the record thread so that 5886 // it is not destroyed while exit() is executed 5887 sp<RecordThread> thread; 5888 { 5889 Mutex::Autolock _l(mLock); 5890 thread = checkRecordThread_l(input); 5891 if (thread == NULL) { 5892 return BAD_VALUE; 5893 } 5894 5895 ALOGV("closeInput() %d", input); 5896 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5897 mRecordThreads.removeItem(input); 5898 } 5899 thread->exit(); 5900 // The thread entity (active unit of execution) is no longer running here, 5901 // but the ThreadBase container still exists. 5902 5903 AudioStreamIn *in = thread->clearInput(); 5904 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5905 // from now on thread->mInput is NULL 5906 in->hwDev->close_input_stream(in->hwDev, in->stream); 5907 delete in; 5908 5909 return NO_ERROR; 5910} 5911 5912status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5913{ 5914 Mutex::Autolock _l(mLock); 5915 MixerThread *dstThread = checkMixerThread_l(output); 5916 if (dstThread == NULL) { 5917 ALOGW("setStreamOutput() bad output id %d", output); 5918 return BAD_VALUE; 5919 } 5920 5921 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5922 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5923 5924 dstThread->setStreamValid(stream, true); 5925 5926 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5927 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5928 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5929 MixerThread *srcThread = (MixerThread *)thread; 5930 srcThread->setStreamValid(stream, false); 5931 srcThread->invalidateTracks(stream); 5932 } 5933 } 5934 5935 return NO_ERROR; 5936} 5937 5938 5939int AudioFlinger::newAudioSessionId() 5940{ 5941 return nextUniqueId(); 5942} 5943 5944void AudioFlinger::acquireAudioSessionId(int audioSession) 5945{ 5946 Mutex::Autolock _l(mLock); 5947 pid_t caller = IPCThreadState::self()->getCallingPid(); 5948 ALOGV("acquiring %d from %d", audioSession, caller); 5949 size_t num = mAudioSessionRefs.size(); 5950 for (size_t i = 0; i< num; i++) { 5951 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5952 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5953 ref->mCnt++; 5954 ALOGV(" incremented refcount to %d", ref->mCnt); 5955 return; 5956 } 5957 } 5958 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5959 ALOGV(" added new entry for %d", audioSession); 5960} 5961 5962void AudioFlinger::releaseAudioSessionId(int audioSession) 5963{ 5964 Mutex::Autolock _l(mLock); 5965 pid_t caller = IPCThreadState::self()->getCallingPid(); 5966 ALOGV("releasing %d from %d", audioSession, caller); 5967 size_t num = mAudioSessionRefs.size(); 5968 for (size_t i = 0; i< num; i++) { 5969 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5970 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5971 ref->mCnt--; 5972 ALOGV(" decremented refcount to %d", ref->mCnt); 5973 if (ref->mCnt == 0) { 5974 mAudioSessionRefs.removeAt(i); 5975 delete ref; 5976 purgeStaleEffects_l(); 5977 } 5978 return; 5979 } 5980 } 5981 ALOGW("session id %d not found for pid %d", audioSession, caller); 5982} 5983 5984void AudioFlinger::purgeStaleEffects_l() { 5985 5986 ALOGV("purging stale effects"); 5987 5988 Vector< sp<EffectChain> > chains; 5989 5990 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5991 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5992 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5993 sp<EffectChain> ec = t->mEffectChains[j]; 5994 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5995 chains.push(ec); 5996 } 5997 } 5998 } 5999 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6000 sp<RecordThread> t = mRecordThreads.valueAt(i); 6001 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6002 sp<EffectChain> ec = t->mEffectChains[j]; 6003 chains.push(ec); 6004 } 6005 } 6006 6007 for (size_t i = 0; i < chains.size(); i++) { 6008 sp<EffectChain> ec = chains[i]; 6009 int sessionid = ec->sessionId(); 6010 sp<ThreadBase> t = ec->mThread.promote(); 6011 if (t == 0) { 6012 continue; 6013 } 6014 size_t numsessionrefs = mAudioSessionRefs.size(); 6015 bool found = false; 6016 for (size_t k = 0; k < numsessionrefs; k++) { 6017 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6018 if (ref->mSessionid == sessionid) { 6019 ALOGV(" session %d still exists for %d with %d refs", 6020 sessionid, ref->mPid, ref->mCnt); 6021 found = true; 6022 break; 6023 } 6024 } 6025 if (!found) { 6026 // remove all effects from the chain 6027 while (ec->mEffects.size()) { 6028 sp<EffectModule> effect = ec->mEffects[0]; 6029 effect->unPin(); 6030 Mutex::Autolock _l (t->mLock); 6031 t->removeEffect_l(effect); 6032 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6033 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6034 if (handle != 0) { 6035 handle->mEffect.clear(); 6036 if (handle->mHasControl && handle->mEnabled) { 6037 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6038 } 6039 } 6040 } 6041 AudioSystem::unregisterEffect(effect->id()); 6042 } 6043 } 6044 } 6045 return; 6046} 6047 6048// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6049AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6050{ 6051 return mPlaybackThreads.valueFor(output).get(); 6052} 6053 6054// checkMixerThread_l() must be called with AudioFlinger::mLock held 6055AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6056{ 6057 PlaybackThread *thread = checkPlaybackThread_l(output); 6058 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6059} 6060 6061// checkRecordThread_l() must be called with AudioFlinger::mLock held 6062AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6063{ 6064 return mRecordThreads.valueFor(input).get(); 6065} 6066 6067uint32_t AudioFlinger::nextUniqueId() 6068{ 6069 return android_atomic_inc(&mNextUniqueId); 6070} 6071 6072AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6073{ 6074 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6075 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6076 AudioStreamOut *output = thread->getOutput(); 6077 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6078 return thread; 6079 } 6080 } 6081 return NULL; 6082} 6083 6084uint32_t AudioFlinger::primaryOutputDevice_l() const 6085{ 6086 PlaybackThread *thread = primaryPlaybackThread_l(); 6087 6088 if (thread == NULL) { 6089 return 0; 6090 } 6091 6092 return thread->device(); 6093} 6094 6095sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6096 int triggerSession, 6097 int listenerSession, 6098 sync_event_callback_t callBack, 6099 void *cookie) 6100{ 6101 Mutex::Autolock _l(mLock); 6102 6103 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6104 status_t playStatus = NAME_NOT_FOUND; 6105 status_t recStatus = NAME_NOT_FOUND; 6106 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6107 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6108 if (playStatus == NO_ERROR) { 6109 return event; 6110 } 6111 } 6112 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6113 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6114 if (recStatus == NO_ERROR) { 6115 return event; 6116 } 6117 } 6118 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6119 mPendingSyncEvents.add(event); 6120 } else { 6121 ALOGV("createSyncEvent() invalid event %d", event->type()); 6122 event.clear(); 6123 } 6124 return event; 6125} 6126 6127// ---------------------------------------------------------------------------- 6128// Effect management 6129// ---------------------------------------------------------------------------- 6130 6131 6132status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6133{ 6134 Mutex::Autolock _l(mLock); 6135 return EffectQueryNumberEffects(numEffects); 6136} 6137 6138status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6139{ 6140 Mutex::Autolock _l(mLock); 6141 return EffectQueryEffect(index, descriptor); 6142} 6143 6144status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6145 effect_descriptor_t *descriptor) const 6146{ 6147 Mutex::Autolock _l(mLock); 6148 return EffectGetDescriptor(pUuid, descriptor); 6149} 6150 6151 6152sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6153 effect_descriptor_t *pDesc, 6154 const sp<IEffectClient>& effectClient, 6155 int32_t priority, 6156 audio_io_handle_t io, 6157 int sessionId, 6158 status_t *status, 6159 int *id, 6160 int *enabled) 6161{ 6162 status_t lStatus = NO_ERROR; 6163 sp<EffectHandle> handle; 6164 effect_descriptor_t desc; 6165 6166 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6167 pid, effectClient.get(), priority, sessionId, io); 6168 6169 if (pDesc == NULL) { 6170 lStatus = BAD_VALUE; 6171 goto Exit; 6172 } 6173 6174 // check audio settings permission for global effects 6175 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6176 lStatus = PERMISSION_DENIED; 6177 goto Exit; 6178 } 6179 6180 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6181 // that can only be created by audio policy manager (running in same process) 6182 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6183 lStatus = PERMISSION_DENIED; 6184 goto Exit; 6185 } 6186 6187 if (io == 0) { 6188 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6189 // output must be specified by AudioPolicyManager when using session 6190 // AUDIO_SESSION_OUTPUT_STAGE 6191 lStatus = BAD_VALUE; 6192 goto Exit; 6193 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6194 // if the output returned by getOutputForEffect() is removed before we lock the 6195 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6196 // and we will exit safely 6197 io = AudioSystem::getOutputForEffect(&desc); 6198 } 6199 } 6200 6201 { 6202 Mutex::Autolock _l(mLock); 6203 6204 6205 if (!EffectIsNullUuid(&pDesc->uuid)) { 6206 // if uuid is specified, request effect descriptor 6207 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6208 if (lStatus < 0) { 6209 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6210 goto Exit; 6211 } 6212 } else { 6213 // if uuid is not specified, look for an available implementation 6214 // of the required type in effect factory 6215 if (EffectIsNullUuid(&pDesc->type)) { 6216 ALOGW("createEffect() no effect type"); 6217 lStatus = BAD_VALUE; 6218 goto Exit; 6219 } 6220 uint32_t numEffects = 0; 6221 effect_descriptor_t d; 6222 d.flags = 0; // prevent compiler warning 6223 bool found = false; 6224 6225 lStatus = EffectQueryNumberEffects(&numEffects); 6226 if (lStatus < 0) { 6227 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6228 goto Exit; 6229 } 6230 for (uint32_t i = 0; i < numEffects; i++) { 6231 lStatus = EffectQueryEffect(i, &desc); 6232 if (lStatus < 0) { 6233 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6234 continue; 6235 } 6236 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6237 // If matching type found save effect descriptor. If the session is 6238 // 0 and the effect is not auxiliary, continue enumeration in case 6239 // an auxiliary version of this effect type is available 6240 found = true; 6241 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6242 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6243 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6244 break; 6245 } 6246 } 6247 } 6248 if (!found) { 6249 lStatus = BAD_VALUE; 6250 ALOGW("createEffect() effect not found"); 6251 goto Exit; 6252 } 6253 // For same effect type, chose auxiliary version over insert version if 6254 // connect to output mix (Compliance to OpenSL ES) 6255 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6256 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6257 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6258 } 6259 } 6260 6261 // Do not allow auxiliary effects on a session different from 0 (output mix) 6262 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6263 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6264 lStatus = INVALID_OPERATION; 6265 goto Exit; 6266 } 6267 6268 // check recording permission for visualizer 6269 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6270 !recordingAllowed()) { 6271 lStatus = PERMISSION_DENIED; 6272 goto Exit; 6273 } 6274 6275 // return effect descriptor 6276 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6277 6278 // If output is not specified try to find a matching audio session ID in one of the 6279 // output threads. 6280 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6281 // because of code checking output when entering the function. 6282 // Note: io is never 0 when creating an effect on an input 6283 if (io == 0) { 6284 // look for the thread where the specified audio session is present 6285 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6286 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6287 io = mPlaybackThreads.keyAt(i); 6288 break; 6289 } 6290 } 6291 if (io == 0) { 6292 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6293 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6294 io = mRecordThreads.keyAt(i); 6295 break; 6296 } 6297 } 6298 } 6299 // If no output thread contains the requested session ID, default to 6300 // first output. The effect chain will be moved to the correct output 6301 // thread when a track with the same session ID is created 6302 if (io == 0 && mPlaybackThreads.size()) { 6303 io = mPlaybackThreads.keyAt(0); 6304 } 6305 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6306 } 6307 ThreadBase *thread = checkRecordThread_l(io); 6308 if (thread == NULL) { 6309 thread = checkPlaybackThread_l(io); 6310 if (thread == NULL) { 6311 ALOGE("createEffect() unknown output thread"); 6312 lStatus = BAD_VALUE; 6313 goto Exit; 6314 } 6315 } 6316 6317 sp<Client> client = registerPid_l(pid); 6318 6319 // create effect on selected output thread 6320 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6321 &desc, enabled, &lStatus); 6322 if (handle != 0 && id != NULL) { 6323 *id = handle->id(); 6324 } 6325 } 6326 6327Exit: 6328 if (status != NULL) { 6329 *status = lStatus; 6330 } 6331 return handle; 6332} 6333 6334status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6335 audio_io_handle_t dstOutput) 6336{ 6337 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6338 sessionId, srcOutput, dstOutput); 6339 Mutex::Autolock _l(mLock); 6340 if (srcOutput == dstOutput) { 6341 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6342 return NO_ERROR; 6343 } 6344 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6345 if (srcThread == NULL) { 6346 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6347 return BAD_VALUE; 6348 } 6349 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6350 if (dstThread == NULL) { 6351 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6352 return BAD_VALUE; 6353 } 6354 6355 Mutex::Autolock _dl(dstThread->mLock); 6356 Mutex::Autolock _sl(srcThread->mLock); 6357 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6358 6359 return NO_ERROR; 6360} 6361 6362// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6363status_t AudioFlinger::moveEffectChain_l(int sessionId, 6364 AudioFlinger::PlaybackThread *srcThread, 6365 AudioFlinger::PlaybackThread *dstThread, 6366 bool reRegister) 6367{ 6368 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6369 sessionId, srcThread, dstThread); 6370 6371 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6372 if (chain == 0) { 6373 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6374 sessionId, srcThread); 6375 return INVALID_OPERATION; 6376 } 6377 6378 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6379 // so that a new chain is created with correct parameters when first effect is added. This is 6380 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6381 // removed. 6382 srcThread->removeEffectChain_l(chain); 6383 6384 // transfer all effects one by one so that new effect chain is created on new thread with 6385 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6386 audio_io_handle_t dstOutput = dstThread->id(); 6387 sp<EffectChain> dstChain; 6388 uint32_t strategy = 0; // prevent compiler warning 6389 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6390 while (effect != 0) { 6391 srcThread->removeEffect_l(effect); 6392 dstThread->addEffect_l(effect); 6393 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6394 if (effect->state() == EffectModule::ACTIVE || 6395 effect->state() == EffectModule::STOPPING) { 6396 effect->start(); 6397 } 6398 // if the move request is not received from audio policy manager, the effect must be 6399 // re-registered with the new strategy and output 6400 if (dstChain == 0) { 6401 dstChain = effect->chain().promote(); 6402 if (dstChain == 0) { 6403 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6404 srcThread->addEffect_l(effect); 6405 return NO_INIT; 6406 } 6407 strategy = dstChain->strategy(); 6408 } 6409 if (reRegister) { 6410 AudioSystem::unregisterEffect(effect->id()); 6411 AudioSystem::registerEffect(&effect->desc(), 6412 dstOutput, 6413 strategy, 6414 sessionId, 6415 effect->id()); 6416 } 6417 effect = chain->getEffectFromId_l(0); 6418 } 6419 6420 return NO_ERROR; 6421} 6422 6423 6424// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6425sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6426 const sp<AudioFlinger::Client>& client, 6427 const sp<IEffectClient>& effectClient, 6428 int32_t priority, 6429 int sessionId, 6430 effect_descriptor_t *desc, 6431 int *enabled, 6432 status_t *status 6433 ) 6434{ 6435 sp<EffectModule> effect; 6436 sp<EffectHandle> handle; 6437 status_t lStatus; 6438 sp<EffectChain> chain; 6439 bool chainCreated = false; 6440 bool effectCreated = false; 6441 bool effectRegistered = false; 6442 6443 lStatus = initCheck(); 6444 if (lStatus != NO_ERROR) { 6445 ALOGW("createEffect_l() Audio driver not initialized."); 6446 goto Exit; 6447 } 6448 6449 // Do not allow effects with session ID 0 on direct output or duplicating threads 6450 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6451 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6452 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6453 desc->name, sessionId); 6454 lStatus = BAD_VALUE; 6455 goto Exit; 6456 } 6457 // Only Pre processor effects are allowed on input threads and only on input threads 6458 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6459 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6460 desc->name, desc->flags, mType); 6461 lStatus = BAD_VALUE; 6462 goto Exit; 6463 } 6464 6465 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6466 6467 { // scope for mLock 6468 Mutex::Autolock _l(mLock); 6469 6470 // check for existing effect chain with the requested audio session 6471 chain = getEffectChain_l(sessionId); 6472 if (chain == 0) { 6473 // create a new chain for this session 6474 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6475 chain = new EffectChain(this, sessionId); 6476 addEffectChain_l(chain); 6477 chain->setStrategy(getStrategyForSession_l(sessionId)); 6478 chainCreated = true; 6479 } else { 6480 effect = chain->getEffectFromDesc_l(desc); 6481 } 6482 6483 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6484 6485 if (effect == 0) { 6486 int id = mAudioFlinger->nextUniqueId(); 6487 // Check CPU and memory usage 6488 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6489 if (lStatus != NO_ERROR) { 6490 goto Exit; 6491 } 6492 effectRegistered = true; 6493 // create a new effect module if none present in the chain 6494 effect = new EffectModule(this, chain, desc, id, sessionId); 6495 lStatus = effect->status(); 6496 if (lStatus != NO_ERROR) { 6497 goto Exit; 6498 } 6499 lStatus = chain->addEffect_l(effect); 6500 if (lStatus != NO_ERROR) { 6501 goto Exit; 6502 } 6503 effectCreated = true; 6504 6505 effect->setDevice(mDevice); 6506 effect->setMode(mAudioFlinger->getMode()); 6507 } 6508 // create effect handle and connect it to effect module 6509 handle = new EffectHandle(effect, client, effectClient, priority); 6510 lStatus = effect->addHandle(handle); 6511 if (enabled != NULL) { 6512 *enabled = (int)effect->isEnabled(); 6513 } 6514 } 6515 6516Exit: 6517 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6518 Mutex::Autolock _l(mLock); 6519 if (effectCreated) { 6520 chain->removeEffect_l(effect); 6521 } 6522 if (effectRegistered) { 6523 AudioSystem::unregisterEffect(effect->id()); 6524 } 6525 if (chainCreated) { 6526 removeEffectChain_l(chain); 6527 } 6528 handle.clear(); 6529 } 6530 6531 if (status != NULL) { 6532 *status = lStatus; 6533 } 6534 return handle; 6535} 6536 6537sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6538{ 6539 sp<EffectChain> chain = getEffectChain_l(sessionId); 6540 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6541} 6542 6543// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6544// PlaybackThread::mLock held 6545status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6546{ 6547 // check for existing effect chain with the requested audio session 6548 int sessionId = effect->sessionId(); 6549 sp<EffectChain> chain = getEffectChain_l(sessionId); 6550 bool chainCreated = false; 6551 6552 if (chain == 0) { 6553 // create a new chain for this session 6554 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6555 chain = new EffectChain(this, sessionId); 6556 addEffectChain_l(chain); 6557 chain->setStrategy(getStrategyForSession_l(sessionId)); 6558 chainCreated = true; 6559 } 6560 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6561 6562 if (chain->getEffectFromId_l(effect->id()) != 0) { 6563 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6564 this, effect->desc().name, chain.get()); 6565 return BAD_VALUE; 6566 } 6567 6568 status_t status = chain->addEffect_l(effect); 6569 if (status != NO_ERROR) { 6570 if (chainCreated) { 6571 removeEffectChain_l(chain); 6572 } 6573 return status; 6574 } 6575 6576 effect->setDevice(mDevice); 6577 effect->setMode(mAudioFlinger->getMode()); 6578 return NO_ERROR; 6579} 6580 6581void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6582 6583 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6584 effect_descriptor_t desc = effect->desc(); 6585 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6586 detachAuxEffect_l(effect->id()); 6587 } 6588 6589 sp<EffectChain> chain = effect->chain().promote(); 6590 if (chain != 0) { 6591 // remove effect chain if removing last effect 6592 if (chain->removeEffect_l(effect) == 0) { 6593 removeEffectChain_l(chain); 6594 } 6595 } else { 6596 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6597 } 6598} 6599 6600void AudioFlinger::ThreadBase::lockEffectChains_l( 6601 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6602{ 6603 effectChains = mEffectChains; 6604 for (size_t i = 0; i < mEffectChains.size(); i++) { 6605 mEffectChains[i]->lock(); 6606 } 6607} 6608 6609void AudioFlinger::ThreadBase::unlockEffectChains( 6610 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6611{ 6612 for (size_t i = 0; i < effectChains.size(); i++) { 6613 effectChains[i]->unlock(); 6614 } 6615} 6616 6617sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6618{ 6619 Mutex::Autolock _l(mLock); 6620 return getEffectChain_l(sessionId); 6621} 6622 6623sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6624{ 6625 size_t size = mEffectChains.size(); 6626 for (size_t i = 0; i < size; i++) { 6627 if (mEffectChains[i]->sessionId() == sessionId) { 6628 return mEffectChains[i]; 6629 } 6630 } 6631 return 0; 6632} 6633 6634void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6635{ 6636 Mutex::Autolock _l(mLock); 6637 size_t size = mEffectChains.size(); 6638 for (size_t i = 0; i < size; i++) { 6639 mEffectChains[i]->setMode_l(mode); 6640 } 6641} 6642 6643void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6644 const wp<EffectHandle>& handle, 6645 bool unpinIfLast) { 6646 6647 Mutex::Autolock _l(mLock); 6648 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6649 // delete the effect module if removing last handle on it 6650 if (effect->removeHandle(handle) == 0) { 6651 if (!effect->isPinned() || unpinIfLast) { 6652 removeEffect_l(effect); 6653 AudioSystem::unregisterEffect(effect->id()); 6654 } 6655 } 6656} 6657 6658status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6659{ 6660 int session = chain->sessionId(); 6661 int16_t *buffer = mMixBuffer; 6662 bool ownsBuffer = false; 6663 6664 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6665 if (session > 0) { 6666 // Only one effect chain can be present in direct output thread and it uses 6667 // the mix buffer as input 6668 if (mType != DIRECT) { 6669 size_t numSamples = mFrameCount * mChannelCount; 6670 buffer = new int16_t[numSamples]; 6671 memset(buffer, 0, numSamples * sizeof(int16_t)); 6672 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6673 ownsBuffer = true; 6674 } 6675 6676 // Attach all tracks with same session ID to this chain. 6677 for (size_t i = 0; i < mTracks.size(); ++i) { 6678 sp<Track> track = mTracks[i]; 6679 if (session == track->sessionId()) { 6680 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6681 track->setMainBuffer(buffer); 6682 chain->incTrackCnt(); 6683 } 6684 } 6685 6686 // indicate all active tracks in the chain 6687 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6688 sp<Track> track = mActiveTracks[i].promote(); 6689 if (track == 0) continue; 6690 if (session == track->sessionId()) { 6691 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6692 chain->incActiveTrackCnt(); 6693 } 6694 } 6695 } 6696 6697 chain->setInBuffer(buffer, ownsBuffer); 6698 chain->setOutBuffer(mMixBuffer); 6699 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6700 // chains list in order to be processed last as it contains output stage effects 6701 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6702 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6703 // after track specific effects and before output stage 6704 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6705 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6706 // Effect chain for other sessions are inserted at beginning of effect 6707 // chains list to be processed before output mix effects. Relative order between other 6708 // sessions is not important 6709 size_t size = mEffectChains.size(); 6710 size_t i = 0; 6711 for (i = 0; i < size; i++) { 6712 if (mEffectChains[i]->sessionId() < session) break; 6713 } 6714 mEffectChains.insertAt(chain, i); 6715 checkSuspendOnAddEffectChain_l(chain); 6716 6717 return NO_ERROR; 6718} 6719 6720size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6721{ 6722 int session = chain->sessionId(); 6723 6724 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6725 6726 for (size_t i = 0; i < mEffectChains.size(); i++) { 6727 if (chain == mEffectChains[i]) { 6728 mEffectChains.removeAt(i); 6729 // detach all active tracks from the chain 6730 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6731 sp<Track> track = mActiveTracks[i].promote(); 6732 if (track == 0) continue; 6733 if (session == track->sessionId()) { 6734 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6735 chain.get(), session); 6736 chain->decActiveTrackCnt(); 6737 } 6738 } 6739 6740 // detach all tracks with same session ID from this chain 6741 for (size_t i = 0; i < mTracks.size(); ++i) { 6742 sp<Track> track = mTracks[i]; 6743 if (session == track->sessionId()) { 6744 track->setMainBuffer(mMixBuffer); 6745 chain->decTrackCnt(); 6746 } 6747 } 6748 break; 6749 } 6750 } 6751 return mEffectChains.size(); 6752} 6753 6754status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6755 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6756{ 6757 Mutex::Autolock _l(mLock); 6758 return attachAuxEffect_l(track, EffectId); 6759} 6760 6761status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6762 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6763{ 6764 status_t status = NO_ERROR; 6765 6766 if (EffectId == 0) { 6767 track->setAuxBuffer(0, NULL); 6768 } else { 6769 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6770 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6771 if (effect != 0) { 6772 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6773 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6774 } else { 6775 status = INVALID_OPERATION; 6776 } 6777 } else { 6778 status = BAD_VALUE; 6779 } 6780 } 6781 return status; 6782} 6783 6784void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6785{ 6786 for (size_t i = 0; i < mTracks.size(); ++i) { 6787 sp<Track> track = mTracks[i]; 6788 if (track->auxEffectId() == effectId) { 6789 attachAuxEffect_l(track, 0); 6790 } 6791 } 6792} 6793 6794status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6795{ 6796 // only one chain per input thread 6797 if (mEffectChains.size() != 0) { 6798 return INVALID_OPERATION; 6799 } 6800 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6801 6802 chain->setInBuffer(NULL); 6803 chain->setOutBuffer(NULL); 6804 6805 checkSuspendOnAddEffectChain_l(chain); 6806 6807 mEffectChains.add(chain); 6808 6809 return NO_ERROR; 6810} 6811 6812size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6813{ 6814 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6815 ALOGW_IF(mEffectChains.size() != 1, 6816 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6817 chain.get(), mEffectChains.size(), this); 6818 if (mEffectChains.size() == 1) { 6819 mEffectChains.removeAt(0); 6820 } 6821 return 0; 6822} 6823 6824// ---------------------------------------------------------------------------- 6825// EffectModule implementation 6826// ---------------------------------------------------------------------------- 6827 6828#undef LOG_TAG 6829#define LOG_TAG "AudioFlinger::EffectModule" 6830 6831AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6832 const wp<AudioFlinger::EffectChain>& chain, 6833 effect_descriptor_t *desc, 6834 int id, 6835 int sessionId) 6836 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6837 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6838{ 6839 ALOGV("Constructor %p", this); 6840 int lStatus; 6841 if (thread == NULL) { 6842 return; 6843 } 6844 6845 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6846 6847 // create effect engine from effect factory 6848 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6849 6850 if (mStatus != NO_ERROR) { 6851 return; 6852 } 6853 lStatus = init(); 6854 if (lStatus < 0) { 6855 mStatus = lStatus; 6856 goto Error; 6857 } 6858 6859 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6860 mPinned = true; 6861 } 6862 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6863 return; 6864Error: 6865 EffectRelease(mEffectInterface); 6866 mEffectInterface = NULL; 6867 ALOGV("Constructor Error %d", mStatus); 6868} 6869 6870AudioFlinger::EffectModule::~EffectModule() 6871{ 6872 ALOGV("Destructor %p", this); 6873 if (mEffectInterface != NULL) { 6874 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6875 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6876 sp<ThreadBase> thread = mThread.promote(); 6877 if (thread != 0) { 6878 audio_stream_t *stream = thread->stream(); 6879 if (stream != NULL) { 6880 stream->remove_audio_effect(stream, mEffectInterface); 6881 } 6882 } 6883 } 6884 // release effect engine 6885 EffectRelease(mEffectInterface); 6886 } 6887} 6888 6889status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6890{ 6891 status_t status; 6892 6893 Mutex::Autolock _l(mLock); 6894 int priority = handle->priority(); 6895 size_t size = mHandles.size(); 6896 sp<EffectHandle> h; 6897 size_t i; 6898 for (i = 0; i < size; i++) { 6899 h = mHandles[i].promote(); 6900 if (h == 0) continue; 6901 if (h->priority() <= priority) break; 6902 } 6903 // if inserted in first place, move effect control from previous owner to this handle 6904 if (i == 0) { 6905 bool enabled = false; 6906 if (h != 0) { 6907 enabled = h->enabled(); 6908 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6909 } 6910 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6911 status = NO_ERROR; 6912 } else { 6913 status = ALREADY_EXISTS; 6914 } 6915 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6916 mHandles.insertAt(handle, i); 6917 return status; 6918} 6919 6920size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6921{ 6922 Mutex::Autolock _l(mLock); 6923 size_t size = mHandles.size(); 6924 size_t i; 6925 for (i = 0; i < size; i++) { 6926 if (mHandles[i] == handle) break; 6927 } 6928 if (i == size) { 6929 return size; 6930 } 6931 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6932 6933 bool enabled = false; 6934 EffectHandle *hdl = handle.unsafe_get(); 6935 if (hdl != NULL) { 6936 ALOGV("removeHandle() unsafe_get OK"); 6937 enabled = hdl->enabled(); 6938 } 6939 mHandles.removeAt(i); 6940 size = mHandles.size(); 6941 // if removed from first place, move effect control from this handle to next in line 6942 if (i == 0 && size != 0) { 6943 sp<EffectHandle> h = mHandles[0].promote(); 6944 if (h != 0) { 6945 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6946 } 6947 } 6948 6949 // Prevent calls to process() and other functions on effect interface from now on. 6950 // The effect engine will be released by the destructor when the last strong reference on 6951 // this object is released which can happen after next process is called. 6952 if (size == 0 && !mPinned) { 6953 mState = DESTROYED; 6954 } 6955 6956 return size; 6957} 6958 6959sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6960{ 6961 Mutex::Autolock _l(mLock); 6962 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6963} 6964 6965void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6966{ 6967 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6968 // keep a strong reference on this EffectModule to avoid calling the 6969 // destructor before we exit 6970 sp<EffectModule> keep(this); 6971 { 6972 sp<ThreadBase> thread = mThread.promote(); 6973 if (thread != 0) { 6974 thread->disconnectEffect(keep, handle, unpinIfLast); 6975 } 6976 } 6977} 6978 6979void AudioFlinger::EffectModule::updateState() { 6980 Mutex::Autolock _l(mLock); 6981 6982 switch (mState) { 6983 case RESTART: 6984 reset_l(); 6985 // FALL THROUGH 6986 6987 case STARTING: 6988 // clear auxiliary effect input buffer for next accumulation 6989 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6990 memset(mConfig.inputCfg.buffer.raw, 6991 0, 6992 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6993 } 6994 start_l(); 6995 mState = ACTIVE; 6996 break; 6997 case STOPPING: 6998 stop_l(); 6999 mDisableWaitCnt = mMaxDisableWaitCnt; 7000 mState = STOPPED; 7001 break; 7002 case STOPPED: 7003 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7004 // turn off sequence. 7005 if (--mDisableWaitCnt == 0) { 7006 reset_l(); 7007 mState = IDLE; 7008 } 7009 break; 7010 default: //IDLE , ACTIVE, DESTROYED 7011 break; 7012 } 7013} 7014 7015void AudioFlinger::EffectModule::process() 7016{ 7017 Mutex::Autolock _l(mLock); 7018 7019 if (mState == DESTROYED || mEffectInterface == NULL || 7020 mConfig.inputCfg.buffer.raw == NULL || 7021 mConfig.outputCfg.buffer.raw == NULL) { 7022 return; 7023 } 7024 7025 if (isProcessEnabled()) { 7026 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7027 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7028 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7029 mConfig.inputCfg.buffer.s32, 7030 mConfig.inputCfg.buffer.frameCount/2); 7031 } 7032 7033 // do the actual processing in the effect engine 7034 int ret = (*mEffectInterface)->process(mEffectInterface, 7035 &mConfig.inputCfg.buffer, 7036 &mConfig.outputCfg.buffer); 7037 7038 // force transition to IDLE state when engine is ready 7039 if (mState == STOPPED && ret == -ENODATA) { 7040 mDisableWaitCnt = 1; 7041 } 7042 7043 // clear auxiliary effect input buffer for next accumulation 7044 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7045 memset(mConfig.inputCfg.buffer.raw, 0, 7046 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7047 } 7048 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7049 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7050 // If an insert effect is idle and input buffer is different from output buffer, 7051 // accumulate input onto output 7052 sp<EffectChain> chain = mChain.promote(); 7053 if (chain != 0 && chain->activeTrackCnt() != 0) { 7054 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7055 int16_t *in = mConfig.inputCfg.buffer.s16; 7056 int16_t *out = mConfig.outputCfg.buffer.s16; 7057 for (size_t i = 0; i < frameCnt; i++) { 7058 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7059 } 7060 } 7061 } 7062} 7063 7064void AudioFlinger::EffectModule::reset_l() 7065{ 7066 if (mEffectInterface == NULL) { 7067 return; 7068 } 7069 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7070} 7071 7072status_t AudioFlinger::EffectModule::configure() 7073{ 7074 uint32_t channels; 7075 if (mEffectInterface == NULL) { 7076 return NO_INIT; 7077 } 7078 7079 sp<ThreadBase> thread = mThread.promote(); 7080 if (thread == 0) { 7081 return DEAD_OBJECT; 7082 } 7083 7084 // TODO: handle configuration of effects replacing track process 7085 if (thread->channelCount() == 1) { 7086 channels = AUDIO_CHANNEL_OUT_MONO; 7087 } else { 7088 channels = AUDIO_CHANNEL_OUT_STEREO; 7089 } 7090 7091 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7092 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7093 } else { 7094 mConfig.inputCfg.channels = channels; 7095 } 7096 mConfig.outputCfg.channels = channels; 7097 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7098 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7099 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7100 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7101 mConfig.inputCfg.bufferProvider.cookie = NULL; 7102 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7103 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7104 mConfig.outputCfg.bufferProvider.cookie = NULL; 7105 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7106 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7107 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7108 // Insert effect: 7109 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7110 // always overwrites output buffer: input buffer == output buffer 7111 // - in other sessions: 7112 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7113 // other effect: overwrites output buffer: input buffer == output buffer 7114 // Auxiliary effect: 7115 // accumulates in output buffer: input buffer != output buffer 7116 // Therefore: accumulate <=> input buffer != output buffer 7117 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7118 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7119 } else { 7120 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7121 } 7122 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7123 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7124 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7125 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7126 7127 ALOGV("configure() %p thread %p buffer %p framecount %d", 7128 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7129 7130 status_t cmdStatus; 7131 uint32_t size = sizeof(int); 7132 status_t status = (*mEffectInterface)->command(mEffectInterface, 7133 EFFECT_CMD_SET_CONFIG, 7134 sizeof(effect_config_t), 7135 &mConfig, 7136 &size, 7137 &cmdStatus); 7138 if (status == 0) { 7139 status = cmdStatus; 7140 } 7141 7142 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7143 (1000 * mConfig.outputCfg.buffer.frameCount); 7144 7145 return status; 7146} 7147 7148status_t AudioFlinger::EffectModule::init() 7149{ 7150 Mutex::Autolock _l(mLock); 7151 if (mEffectInterface == NULL) { 7152 return NO_INIT; 7153 } 7154 status_t cmdStatus; 7155 uint32_t size = sizeof(status_t); 7156 status_t status = (*mEffectInterface)->command(mEffectInterface, 7157 EFFECT_CMD_INIT, 7158 0, 7159 NULL, 7160 &size, 7161 &cmdStatus); 7162 if (status == 0) { 7163 status = cmdStatus; 7164 } 7165 return status; 7166} 7167 7168status_t AudioFlinger::EffectModule::start() 7169{ 7170 Mutex::Autolock _l(mLock); 7171 return start_l(); 7172} 7173 7174status_t AudioFlinger::EffectModule::start_l() 7175{ 7176 if (mEffectInterface == NULL) { 7177 return NO_INIT; 7178 } 7179 status_t cmdStatus; 7180 uint32_t size = sizeof(status_t); 7181 status_t status = (*mEffectInterface)->command(mEffectInterface, 7182 EFFECT_CMD_ENABLE, 7183 0, 7184 NULL, 7185 &size, 7186 &cmdStatus); 7187 if (status == 0) { 7188 status = cmdStatus; 7189 } 7190 if (status == 0 && 7191 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7192 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7193 sp<ThreadBase> thread = mThread.promote(); 7194 if (thread != 0) { 7195 audio_stream_t *stream = thread->stream(); 7196 if (stream != NULL) { 7197 stream->add_audio_effect(stream, mEffectInterface); 7198 } 7199 } 7200 } 7201 return status; 7202} 7203 7204status_t AudioFlinger::EffectModule::stop() 7205{ 7206 Mutex::Autolock _l(mLock); 7207 return stop_l(); 7208} 7209 7210status_t AudioFlinger::EffectModule::stop_l() 7211{ 7212 if (mEffectInterface == NULL) { 7213 return NO_INIT; 7214 } 7215 status_t cmdStatus; 7216 uint32_t size = sizeof(status_t); 7217 status_t status = (*mEffectInterface)->command(mEffectInterface, 7218 EFFECT_CMD_DISABLE, 7219 0, 7220 NULL, 7221 &size, 7222 &cmdStatus); 7223 if (status == 0) { 7224 status = cmdStatus; 7225 } 7226 if (status == 0 && 7227 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7228 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7229 sp<ThreadBase> thread = mThread.promote(); 7230 if (thread != 0) { 7231 audio_stream_t *stream = thread->stream(); 7232 if (stream != NULL) { 7233 stream->remove_audio_effect(stream, mEffectInterface); 7234 } 7235 } 7236 } 7237 return status; 7238} 7239 7240status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7241 uint32_t cmdSize, 7242 void *pCmdData, 7243 uint32_t *replySize, 7244 void *pReplyData) 7245{ 7246 Mutex::Autolock _l(mLock); 7247// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7248 7249 if (mState == DESTROYED || mEffectInterface == NULL) { 7250 return NO_INIT; 7251 } 7252 status_t status = (*mEffectInterface)->command(mEffectInterface, 7253 cmdCode, 7254 cmdSize, 7255 pCmdData, 7256 replySize, 7257 pReplyData); 7258 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7259 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7260 for (size_t i = 1; i < mHandles.size(); i++) { 7261 sp<EffectHandle> h = mHandles[i].promote(); 7262 if (h != 0) { 7263 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7264 } 7265 } 7266 } 7267 return status; 7268} 7269 7270status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7271{ 7272 7273 Mutex::Autolock _l(mLock); 7274 ALOGV("setEnabled %p enabled %d", this, enabled); 7275 7276 if (enabled != isEnabled()) { 7277 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7278 if (enabled && status != NO_ERROR) { 7279 return status; 7280 } 7281 7282 switch (mState) { 7283 // going from disabled to enabled 7284 case IDLE: 7285 mState = STARTING; 7286 break; 7287 case STOPPED: 7288 mState = RESTART; 7289 break; 7290 case STOPPING: 7291 mState = ACTIVE; 7292 break; 7293 7294 // going from enabled to disabled 7295 case RESTART: 7296 mState = STOPPED; 7297 break; 7298 case STARTING: 7299 mState = IDLE; 7300 break; 7301 case ACTIVE: 7302 mState = STOPPING; 7303 break; 7304 case DESTROYED: 7305 return NO_ERROR; // simply ignore as we are being destroyed 7306 } 7307 for (size_t i = 1; i < mHandles.size(); i++) { 7308 sp<EffectHandle> h = mHandles[i].promote(); 7309 if (h != 0) { 7310 h->setEnabled(enabled); 7311 } 7312 } 7313 } 7314 return NO_ERROR; 7315} 7316 7317bool AudioFlinger::EffectModule::isEnabled() const 7318{ 7319 switch (mState) { 7320 case RESTART: 7321 case STARTING: 7322 case ACTIVE: 7323 return true; 7324 case IDLE: 7325 case STOPPING: 7326 case STOPPED: 7327 case DESTROYED: 7328 default: 7329 return false; 7330 } 7331} 7332 7333bool AudioFlinger::EffectModule::isProcessEnabled() const 7334{ 7335 switch (mState) { 7336 case RESTART: 7337 case ACTIVE: 7338 case STOPPING: 7339 case STOPPED: 7340 return true; 7341 case IDLE: 7342 case STARTING: 7343 case DESTROYED: 7344 default: 7345 return false; 7346 } 7347} 7348 7349status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7350{ 7351 Mutex::Autolock _l(mLock); 7352 status_t status = NO_ERROR; 7353 7354 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7355 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7356 if (isProcessEnabled() && 7357 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7358 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7359 status_t cmdStatus; 7360 uint32_t volume[2]; 7361 uint32_t *pVolume = NULL; 7362 uint32_t size = sizeof(volume); 7363 volume[0] = *left; 7364 volume[1] = *right; 7365 if (controller) { 7366 pVolume = volume; 7367 } 7368 status = (*mEffectInterface)->command(mEffectInterface, 7369 EFFECT_CMD_SET_VOLUME, 7370 size, 7371 volume, 7372 &size, 7373 pVolume); 7374 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7375 *left = volume[0]; 7376 *right = volume[1]; 7377 } 7378 } 7379 return status; 7380} 7381 7382status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7383{ 7384 Mutex::Autolock _l(mLock); 7385 status_t status = NO_ERROR; 7386 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7387 // audio pre processing modules on RecordThread can receive both output and 7388 // input device indication in the same call 7389 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7390 if (dev) { 7391 status_t cmdStatus; 7392 uint32_t size = sizeof(status_t); 7393 7394 status = (*mEffectInterface)->command(mEffectInterface, 7395 EFFECT_CMD_SET_DEVICE, 7396 sizeof(uint32_t), 7397 &dev, 7398 &size, 7399 &cmdStatus); 7400 if (status == NO_ERROR) { 7401 status = cmdStatus; 7402 } 7403 } 7404 dev = device & AUDIO_DEVICE_IN_ALL; 7405 if (dev) { 7406 status_t cmdStatus; 7407 uint32_t size = sizeof(status_t); 7408 7409 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7410 EFFECT_CMD_SET_INPUT_DEVICE, 7411 sizeof(uint32_t), 7412 &dev, 7413 &size, 7414 &cmdStatus); 7415 if (status2 == NO_ERROR) { 7416 status2 = cmdStatus; 7417 } 7418 if (status == NO_ERROR) { 7419 status = status2; 7420 } 7421 } 7422 } 7423 return status; 7424} 7425 7426status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7427{ 7428 Mutex::Autolock _l(mLock); 7429 status_t status = NO_ERROR; 7430 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7431 status_t cmdStatus; 7432 uint32_t size = sizeof(status_t); 7433 status = (*mEffectInterface)->command(mEffectInterface, 7434 EFFECT_CMD_SET_AUDIO_MODE, 7435 sizeof(audio_mode_t), 7436 &mode, 7437 &size, 7438 &cmdStatus); 7439 if (status == NO_ERROR) { 7440 status = cmdStatus; 7441 } 7442 } 7443 return status; 7444} 7445 7446void AudioFlinger::EffectModule::setSuspended(bool suspended) 7447{ 7448 Mutex::Autolock _l(mLock); 7449 mSuspended = suspended; 7450} 7451 7452bool AudioFlinger::EffectModule::suspended() const 7453{ 7454 Mutex::Autolock _l(mLock); 7455 return mSuspended; 7456} 7457 7458status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7459{ 7460 const size_t SIZE = 256; 7461 char buffer[SIZE]; 7462 String8 result; 7463 7464 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7465 result.append(buffer); 7466 7467 bool locked = tryLock(mLock); 7468 // failed to lock - AudioFlinger is probably deadlocked 7469 if (!locked) { 7470 result.append("\t\tCould not lock Fx mutex:\n"); 7471 } 7472 7473 result.append("\t\tSession Status State Engine:\n"); 7474 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7475 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7476 result.append(buffer); 7477 7478 result.append("\t\tDescriptor:\n"); 7479 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7480 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7481 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7482 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7483 result.append(buffer); 7484 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7485 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7486 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7487 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7488 result.append(buffer); 7489 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7490 mDescriptor.apiVersion, 7491 mDescriptor.flags); 7492 result.append(buffer); 7493 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7494 mDescriptor.name); 7495 result.append(buffer); 7496 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7497 mDescriptor.implementor); 7498 result.append(buffer); 7499 7500 result.append("\t\t- Input configuration:\n"); 7501 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7502 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7503 (uint32_t)mConfig.inputCfg.buffer.raw, 7504 mConfig.inputCfg.buffer.frameCount, 7505 mConfig.inputCfg.samplingRate, 7506 mConfig.inputCfg.channels, 7507 mConfig.inputCfg.format); 7508 result.append(buffer); 7509 7510 result.append("\t\t- Output configuration:\n"); 7511 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7512 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7513 (uint32_t)mConfig.outputCfg.buffer.raw, 7514 mConfig.outputCfg.buffer.frameCount, 7515 mConfig.outputCfg.samplingRate, 7516 mConfig.outputCfg.channels, 7517 mConfig.outputCfg.format); 7518 result.append(buffer); 7519 7520 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7521 result.append(buffer); 7522 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7523 for (size_t i = 0; i < mHandles.size(); ++i) { 7524 sp<EffectHandle> handle = mHandles[i].promote(); 7525 if (handle != 0) { 7526 handle->dump(buffer, SIZE); 7527 result.append(buffer); 7528 } 7529 } 7530 7531 result.append("\n"); 7532 7533 write(fd, result.string(), result.length()); 7534 7535 if (locked) { 7536 mLock.unlock(); 7537 } 7538 7539 return NO_ERROR; 7540} 7541 7542// ---------------------------------------------------------------------------- 7543// EffectHandle implementation 7544// ---------------------------------------------------------------------------- 7545 7546#undef LOG_TAG 7547#define LOG_TAG "AudioFlinger::EffectHandle" 7548 7549AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7550 const sp<AudioFlinger::Client>& client, 7551 const sp<IEffectClient>& effectClient, 7552 int32_t priority) 7553 : BnEffect(), 7554 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7555 mPriority(priority), mHasControl(false), mEnabled(false) 7556{ 7557 ALOGV("constructor %p", this); 7558 7559 if (client == 0) { 7560 return; 7561 } 7562 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7563 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7564 if (mCblkMemory != 0) { 7565 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7566 7567 if (mCblk != NULL) { 7568 new(mCblk) effect_param_cblk_t(); 7569 mBuffer = (uint8_t *)mCblk + bufOffset; 7570 } 7571 } else { 7572 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7573 return; 7574 } 7575} 7576 7577AudioFlinger::EffectHandle::~EffectHandle() 7578{ 7579 ALOGV("Destructor %p", this); 7580 disconnect(false); 7581 ALOGV("Destructor DONE %p", this); 7582} 7583 7584status_t AudioFlinger::EffectHandle::enable() 7585{ 7586 ALOGV("enable %p", this); 7587 if (!mHasControl) return INVALID_OPERATION; 7588 if (mEffect == 0) return DEAD_OBJECT; 7589 7590 if (mEnabled) { 7591 return NO_ERROR; 7592 } 7593 7594 mEnabled = true; 7595 7596 sp<ThreadBase> thread = mEffect->thread().promote(); 7597 if (thread != 0) { 7598 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7599 } 7600 7601 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7602 if (mEffect->suspended()) { 7603 return NO_ERROR; 7604 } 7605 7606 status_t status = mEffect->setEnabled(true); 7607 if (status != NO_ERROR) { 7608 if (thread != 0) { 7609 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7610 } 7611 mEnabled = false; 7612 } 7613 return status; 7614} 7615 7616status_t AudioFlinger::EffectHandle::disable() 7617{ 7618 ALOGV("disable %p", this); 7619 if (!mHasControl) return INVALID_OPERATION; 7620 if (mEffect == 0) return DEAD_OBJECT; 7621 7622 if (!mEnabled) { 7623 return NO_ERROR; 7624 } 7625 mEnabled = false; 7626 7627 if (mEffect->suspended()) { 7628 return NO_ERROR; 7629 } 7630 7631 status_t status = mEffect->setEnabled(false); 7632 7633 sp<ThreadBase> thread = mEffect->thread().promote(); 7634 if (thread != 0) { 7635 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7636 } 7637 7638 return status; 7639} 7640 7641void AudioFlinger::EffectHandle::disconnect() 7642{ 7643 disconnect(true); 7644} 7645 7646void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7647{ 7648 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7649 if (mEffect == 0) { 7650 return; 7651 } 7652 mEffect->disconnect(this, unpinIfLast); 7653 7654 if (mHasControl && mEnabled) { 7655 sp<ThreadBase> thread = mEffect->thread().promote(); 7656 if (thread != 0) { 7657 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7658 } 7659 } 7660 7661 // release sp on module => module destructor can be called now 7662 mEffect.clear(); 7663 if (mClient != 0) { 7664 if (mCblk != NULL) { 7665 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7666 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7667 } 7668 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7669 // Client destructor must run with AudioFlinger mutex locked 7670 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7671 mClient.clear(); 7672 } 7673} 7674 7675status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7676 uint32_t cmdSize, 7677 void *pCmdData, 7678 uint32_t *replySize, 7679 void *pReplyData) 7680{ 7681// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7682// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7683 7684 // only get parameter command is permitted for applications not controlling the effect 7685 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7686 return INVALID_OPERATION; 7687 } 7688 if (mEffect == 0) return DEAD_OBJECT; 7689 if (mClient == 0) return INVALID_OPERATION; 7690 7691 // handle commands that are not forwarded transparently to effect engine 7692 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7693 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7694 // no risk to block the whole media server process or mixer threads is we are stuck here 7695 Mutex::Autolock _l(mCblk->lock); 7696 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7697 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7698 mCblk->serverIndex = 0; 7699 mCblk->clientIndex = 0; 7700 return BAD_VALUE; 7701 } 7702 status_t status = NO_ERROR; 7703 while (mCblk->serverIndex < mCblk->clientIndex) { 7704 int reply; 7705 uint32_t rsize = sizeof(int); 7706 int *p = (int *)(mBuffer + mCblk->serverIndex); 7707 int size = *p++; 7708 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7709 ALOGW("command(): invalid parameter block size"); 7710 break; 7711 } 7712 effect_param_t *param = (effect_param_t *)p; 7713 if (param->psize == 0 || param->vsize == 0) { 7714 ALOGW("command(): null parameter or value size"); 7715 mCblk->serverIndex += size; 7716 continue; 7717 } 7718 uint32_t psize = sizeof(effect_param_t) + 7719 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7720 param->vsize; 7721 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7722 psize, 7723 p, 7724 &rsize, 7725 &reply); 7726 // stop at first error encountered 7727 if (ret != NO_ERROR) { 7728 status = ret; 7729 *(int *)pReplyData = reply; 7730 break; 7731 } else if (reply != NO_ERROR) { 7732 *(int *)pReplyData = reply; 7733 break; 7734 } 7735 mCblk->serverIndex += size; 7736 } 7737 mCblk->serverIndex = 0; 7738 mCblk->clientIndex = 0; 7739 return status; 7740 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7741 *(int *)pReplyData = NO_ERROR; 7742 return enable(); 7743 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7744 *(int *)pReplyData = NO_ERROR; 7745 return disable(); 7746 } 7747 7748 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7749} 7750 7751void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7752{ 7753 ALOGV("setControl %p control %d", this, hasControl); 7754 7755 mHasControl = hasControl; 7756 mEnabled = enabled; 7757 7758 if (signal && mEffectClient != 0) { 7759 mEffectClient->controlStatusChanged(hasControl); 7760 } 7761} 7762 7763void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7764 uint32_t cmdSize, 7765 void *pCmdData, 7766 uint32_t replySize, 7767 void *pReplyData) 7768{ 7769 if (mEffectClient != 0) { 7770 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7771 } 7772} 7773 7774 7775 7776void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7777{ 7778 if (mEffectClient != 0) { 7779 mEffectClient->enableStatusChanged(enabled); 7780 } 7781} 7782 7783status_t AudioFlinger::EffectHandle::onTransact( 7784 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7785{ 7786 return BnEffect::onTransact(code, data, reply, flags); 7787} 7788 7789 7790void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7791{ 7792 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7793 7794 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7795 (mClient == 0) ? getpid_cached : mClient->pid(), 7796 mPriority, 7797 mHasControl, 7798 !locked, 7799 mCblk ? mCblk->clientIndex : 0, 7800 mCblk ? mCblk->serverIndex : 0 7801 ); 7802 7803 if (locked) { 7804 mCblk->lock.unlock(); 7805 } 7806} 7807 7808#undef LOG_TAG 7809#define LOG_TAG "AudioFlinger::EffectChain" 7810 7811AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7812 int sessionId) 7813 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7814 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7815 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7816{ 7817 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7818 if (thread == NULL) { 7819 return; 7820 } 7821 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7822 thread->frameCount(); 7823} 7824 7825AudioFlinger::EffectChain::~EffectChain() 7826{ 7827 if (mOwnInBuffer) { 7828 delete mInBuffer; 7829 } 7830 7831} 7832 7833// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7834sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7835{ 7836 size_t size = mEffects.size(); 7837 7838 for (size_t i = 0; i < size; i++) { 7839 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7840 return mEffects[i]; 7841 } 7842 } 7843 return 0; 7844} 7845 7846// getEffectFromId_l() must be called with ThreadBase::mLock held 7847sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7848{ 7849 size_t size = mEffects.size(); 7850 7851 for (size_t i = 0; i < size; i++) { 7852 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7853 if (id == 0 || mEffects[i]->id() == id) { 7854 return mEffects[i]; 7855 } 7856 } 7857 return 0; 7858} 7859 7860// getEffectFromType_l() must be called with ThreadBase::mLock held 7861sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7862 const effect_uuid_t *type) 7863{ 7864 size_t size = mEffects.size(); 7865 7866 for (size_t i = 0; i < size; i++) { 7867 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7868 return mEffects[i]; 7869 } 7870 } 7871 return 0; 7872} 7873 7874// Must be called with EffectChain::mLock locked 7875void AudioFlinger::EffectChain::process_l() 7876{ 7877 sp<ThreadBase> thread = mThread.promote(); 7878 if (thread == 0) { 7879 ALOGW("process_l(): cannot promote mixer thread"); 7880 return; 7881 } 7882 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7883 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7884 // always process effects unless no more tracks are on the session and the effect tail 7885 // has been rendered 7886 bool doProcess = true; 7887 if (!isGlobalSession) { 7888 bool tracksOnSession = (trackCnt() != 0); 7889 7890 if (!tracksOnSession && mTailBufferCount == 0) { 7891 doProcess = false; 7892 } 7893 7894 if (activeTrackCnt() == 0) { 7895 // if no track is active and the effect tail has not been rendered, 7896 // the input buffer must be cleared here as the mixer process will not do it 7897 if (tracksOnSession || mTailBufferCount > 0) { 7898 size_t numSamples = thread->frameCount() * thread->channelCount(); 7899 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7900 if (mTailBufferCount > 0) { 7901 mTailBufferCount--; 7902 } 7903 } 7904 } 7905 } 7906 7907 size_t size = mEffects.size(); 7908 if (doProcess) { 7909 for (size_t i = 0; i < size; i++) { 7910 mEffects[i]->process(); 7911 } 7912 } 7913 for (size_t i = 0; i < size; i++) { 7914 mEffects[i]->updateState(); 7915 } 7916} 7917 7918// addEffect_l() must be called with PlaybackThread::mLock held 7919status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7920{ 7921 effect_descriptor_t desc = effect->desc(); 7922 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7923 7924 Mutex::Autolock _l(mLock); 7925 effect->setChain(this); 7926 sp<ThreadBase> thread = mThread.promote(); 7927 if (thread == 0) { 7928 return NO_INIT; 7929 } 7930 effect->setThread(thread); 7931 7932 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7933 // Auxiliary effects are inserted at the beginning of mEffects vector as 7934 // they are processed first and accumulated in chain input buffer 7935 mEffects.insertAt(effect, 0); 7936 7937 // the input buffer for auxiliary effect contains mono samples in 7938 // 32 bit format. This is to avoid saturation in AudoMixer 7939 // accumulation stage. Saturation is done in EffectModule::process() before 7940 // calling the process in effect engine 7941 size_t numSamples = thread->frameCount(); 7942 int32_t *buffer = new int32_t[numSamples]; 7943 memset(buffer, 0, numSamples * sizeof(int32_t)); 7944 effect->setInBuffer((int16_t *)buffer); 7945 // auxiliary effects output samples to chain input buffer for further processing 7946 // by insert effects 7947 effect->setOutBuffer(mInBuffer); 7948 } else { 7949 // Insert effects are inserted at the end of mEffects vector as they are processed 7950 // after track and auxiliary effects. 7951 // Insert effect order as a function of indicated preference: 7952 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7953 // another effect is present 7954 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7955 // last effect claiming first position 7956 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7957 // first effect claiming last position 7958 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7959 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7960 // already present 7961 7962 size_t size = mEffects.size(); 7963 size_t idx_insert = size; 7964 ssize_t idx_insert_first = -1; 7965 ssize_t idx_insert_last = -1; 7966 7967 for (size_t i = 0; i < size; i++) { 7968 effect_descriptor_t d = mEffects[i]->desc(); 7969 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7970 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7971 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7972 // check invalid effect chaining combinations 7973 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7974 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7975 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7976 return INVALID_OPERATION; 7977 } 7978 // remember position of first insert effect and by default 7979 // select this as insert position for new effect 7980 if (idx_insert == size) { 7981 idx_insert = i; 7982 } 7983 // remember position of last insert effect claiming 7984 // first position 7985 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7986 idx_insert_first = i; 7987 } 7988 // remember position of first insert effect claiming 7989 // last position 7990 if (iPref == EFFECT_FLAG_INSERT_LAST && 7991 idx_insert_last == -1) { 7992 idx_insert_last = i; 7993 } 7994 } 7995 } 7996 7997 // modify idx_insert from first position if needed 7998 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7999 if (idx_insert_last != -1) { 8000 idx_insert = idx_insert_last; 8001 } else { 8002 idx_insert = size; 8003 } 8004 } else { 8005 if (idx_insert_first != -1) { 8006 idx_insert = idx_insert_first + 1; 8007 } 8008 } 8009 8010 // always read samples from chain input buffer 8011 effect->setInBuffer(mInBuffer); 8012 8013 // if last effect in the chain, output samples to chain 8014 // output buffer, otherwise to chain input buffer 8015 if (idx_insert == size) { 8016 if (idx_insert != 0) { 8017 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8018 mEffects[idx_insert-1]->configure(); 8019 } 8020 effect->setOutBuffer(mOutBuffer); 8021 } else { 8022 effect->setOutBuffer(mInBuffer); 8023 } 8024 mEffects.insertAt(effect, idx_insert); 8025 8026 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8027 } 8028 effect->configure(); 8029 return NO_ERROR; 8030} 8031 8032// removeEffect_l() must be called with PlaybackThread::mLock held 8033size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8034{ 8035 Mutex::Autolock _l(mLock); 8036 size_t size = mEffects.size(); 8037 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8038 8039 for (size_t i = 0; i < size; i++) { 8040 if (effect == mEffects[i]) { 8041 // calling stop here will remove pre-processing effect from the audio HAL. 8042 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8043 // the middle of a read from audio HAL 8044 if (mEffects[i]->state() == EffectModule::ACTIVE || 8045 mEffects[i]->state() == EffectModule::STOPPING) { 8046 mEffects[i]->stop(); 8047 } 8048 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8049 delete[] effect->inBuffer(); 8050 } else { 8051 if (i == size - 1 && i != 0) { 8052 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8053 mEffects[i - 1]->configure(); 8054 } 8055 } 8056 mEffects.removeAt(i); 8057 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8058 break; 8059 } 8060 } 8061 8062 return mEffects.size(); 8063} 8064 8065// setDevice_l() must be called with PlaybackThread::mLock held 8066void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8067{ 8068 size_t size = mEffects.size(); 8069 for (size_t i = 0; i < size; i++) { 8070 mEffects[i]->setDevice(device); 8071 } 8072} 8073 8074// setMode_l() must be called with PlaybackThread::mLock held 8075void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8076{ 8077 size_t size = mEffects.size(); 8078 for (size_t i = 0; i < size; i++) { 8079 mEffects[i]->setMode(mode); 8080 } 8081} 8082 8083// setVolume_l() must be called with PlaybackThread::mLock held 8084bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8085{ 8086 uint32_t newLeft = *left; 8087 uint32_t newRight = *right; 8088 bool hasControl = false; 8089 int ctrlIdx = -1; 8090 size_t size = mEffects.size(); 8091 8092 // first update volume controller 8093 for (size_t i = size; i > 0; i--) { 8094 if (mEffects[i - 1]->isProcessEnabled() && 8095 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8096 ctrlIdx = i - 1; 8097 hasControl = true; 8098 break; 8099 } 8100 } 8101 8102 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8103 if (hasControl) { 8104 *left = mNewLeftVolume; 8105 *right = mNewRightVolume; 8106 } 8107 return hasControl; 8108 } 8109 8110 mVolumeCtrlIdx = ctrlIdx; 8111 mLeftVolume = newLeft; 8112 mRightVolume = newRight; 8113 8114 // second get volume update from volume controller 8115 if (ctrlIdx >= 0) { 8116 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8117 mNewLeftVolume = newLeft; 8118 mNewRightVolume = newRight; 8119 } 8120 // then indicate volume to all other effects in chain. 8121 // Pass altered volume to effects before volume controller 8122 // and requested volume to effects after controller 8123 uint32_t lVol = newLeft; 8124 uint32_t rVol = newRight; 8125 8126 for (size_t i = 0; i < size; i++) { 8127 if ((int)i == ctrlIdx) continue; 8128 // this also works for ctrlIdx == -1 when there is no volume controller 8129 if ((int)i > ctrlIdx) { 8130 lVol = *left; 8131 rVol = *right; 8132 } 8133 mEffects[i]->setVolume(&lVol, &rVol, false); 8134 } 8135 *left = newLeft; 8136 *right = newRight; 8137 8138 return hasControl; 8139} 8140 8141status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8142{ 8143 const size_t SIZE = 256; 8144 char buffer[SIZE]; 8145 String8 result; 8146 8147 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8148 result.append(buffer); 8149 8150 bool locked = tryLock(mLock); 8151 // failed to lock - AudioFlinger is probably deadlocked 8152 if (!locked) { 8153 result.append("\tCould not lock mutex:\n"); 8154 } 8155 8156 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8157 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8158 mEffects.size(), 8159 (uint32_t)mInBuffer, 8160 (uint32_t)mOutBuffer, 8161 mActiveTrackCnt); 8162 result.append(buffer); 8163 write(fd, result.string(), result.size()); 8164 8165 for (size_t i = 0; i < mEffects.size(); ++i) { 8166 sp<EffectModule> effect = mEffects[i]; 8167 if (effect != 0) { 8168 effect->dump(fd, args); 8169 } 8170 } 8171 8172 if (locked) { 8173 mLock.unlock(); 8174 } 8175 8176 return NO_ERROR; 8177} 8178 8179// must be called with ThreadBase::mLock held 8180void AudioFlinger::EffectChain::setEffectSuspended_l( 8181 const effect_uuid_t *type, bool suspend) 8182{ 8183 sp<SuspendedEffectDesc> desc; 8184 // use effect type UUID timelow as key as there is no real risk of identical 8185 // timeLow fields among effect type UUIDs. 8186 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8187 if (suspend) { 8188 if (index >= 0) { 8189 desc = mSuspendedEffects.valueAt(index); 8190 } else { 8191 desc = new SuspendedEffectDesc(); 8192 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8193 mSuspendedEffects.add(type->timeLow, desc); 8194 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8195 } 8196 if (desc->mRefCount++ == 0) { 8197 sp<EffectModule> effect = getEffectIfEnabled(type); 8198 if (effect != 0) { 8199 desc->mEffect = effect; 8200 effect->setSuspended(true); 8201 effect->setEnabled(false); 8202 } 8203 } 8204 } else { 8205 if (index < 0) { 8206 return; 8207 } 8208 desc = mSuspendedEffects.valueAt(index); 8209 if (desc->mRefCount <= 0) { 8210 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8211 desc->mRefCount = 1; 8212 } 8213 if (--desc->mRefCount == 0) { 8214 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8215 if (desc->mEffect != 0) { 8216 sp<EffectModule> effect = desc->mEffect.promote(); 8217 if (effect != 0) { 8218 effect->setSuspended(false); 8219 sp<EffectHandle> handle = effect->controlHandle(); 8220 if (handle != 0) { 8221 effect->setEnabled(handle->enabled()); 8222 } 8223 } 8224 desc->mEffect.clear(); 8225 } 8226 mSuspendedEffects.removeItemsAt(index); 8227 } 8228 } 8229} 8230 8231// must be called with ThreadBase::mLock held 8232void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8233{ 8234 sp<SuspendedEffectDesc> desc; 8235 8236 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8237 if (suspend) { 8238 if (index >= 0) { 8239 desc = mSuspendedEffects.valueAt(index); 8240 } else { 8241 desc = new SuspendedEffectDesc(); 8242 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8243 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8244 } 8245 if (desc->mRefCount++ == 0) { 8246 Vector< sp<EffectModule> > effects; 8247 getSuspendEligibleEffects(effects); 8248 for (size_t i = 0; i < effects.size(); i++) { 8249 setEffectSuspended_l(&effects[i]->desc().type, true); 8250 } 8251 } 8252 } else { 8253 if (index < 0) { 8254 return; 8255 } 8256 desc = mSuspendedEffects.valueAt(index); 8257 if (desc->mRefCount <= 0) { 8258 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8259 desc->mRefCount = 1; 8260 } 8261 if (--desc->mRefCount == 0) { 8262 Vector<const effect_uuid_t *> types; 8263 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8264 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8265 continue; 8266 } 8267 types.add(&mSuspendedEffects.valueAt(i)->mType); 8268 } 8269 for (size_t i = 0; i < types.size(); i++) { 8270 setEffectSuspended_l(types[i], false); 8271 } 8272 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8273 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8274 } 8275 } 8276} 8277 8278 8279// The volume effect is used for automated tests only 8280#ifndef OPENSL_ES_H_ 8281static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8282 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8283const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8284#endif //OPENSL_ES_H_ 8285 8286bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8287{ 8288 // auxiliary effects and visualizer are never suspended on output mix 8289 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8290 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8291 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8292 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8293 return false; 8294 } 8295 return true; 8296} 8297 8298void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8299{ 8300 effects.clear(); 8301 for (size_t i = 0; i < mEffects.size(); i++) { 8302 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8303 effects.add(mEffects[i]); 8304 } 8305 } 8306} 8307 8308sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8309 const effect_uuid_t *type) 8310{ 8311 sp<EffectModule> effect = getEffectFromType_l(type); 8312 return effect != 0 && effect->isEnabled() ? effect : 0; 8313} 8314 8315void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8316 bool enabled) 8317{ 8318 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8319 if (enabled) { 8320 if (index < 0) { 8321 // if the effect is not suspend check if all effects are suspended 8322 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8323 if (index < 0) { 8324 return; 8325 } 8326 if (!isEffectEligibleForSuspend(effect->desc())) { 8327 return; 8328 } 8329 setEffectSuspended_l(&effect->desc().type, enabled); 8330 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8331 if (index < 0) { 8332 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8333 return; 8334 } 8335 } 8336 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8337 effect->desc().type.timeLow); 8338 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8339 // if effect is requested to suspended but was not yet enabled, supend it now. 8340 if (desc->mEffect == 0) { 8341 desc->mEffect = effect; 8342 effect->setEnabled(false); 8343 effect->setSuspended(true); 8344 } 8345 } else { 8346 if (index < 0) { 8347 return; 8348 } 8349 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8350 effect->desc().type.timeLow); 8351 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8352 desc->mEffect.clear(); 8353 effect->setSuspended(false); 8354 } 8355} 8356 8357#undef LOG_TAG 8358#define LOG_TAG "AudioFlinger" 8359 8360// ---------------------------------------------------------------------------- 8361 8362status_t AudioFlinger::onTransact( 8363 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8364{ 8365 return BnAudioFlinger::onTransact(code, data, reply, flags); 8366} 8367 8368}; // namespace android 8369