AudioFlinger.cpp revision b83d38feeeb88a8a2a6219e1fca2480b5a14fb0d
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 if (lStatus == NO_ERROR) { 518 trackHandle = new TrackHandle(track); 519 } else { 520 // remove local strong reference to Client before deleting the Track so that the Client 521 // destructor is called by the TrackBase destructor with mLock held 522 client.clear(); 523 track.clear(); 524 } 525 526Exit: 527 if (status != NULL) { 528 *status = lStatus; 529 } 530 return trackHandle; 531} 532 533uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 534{ 535 Mutex::Autolock _l(mLock); 536 PlaybackThread *thread = checkPlaybackThread_l(output); 537 if (thread == NULL) { 538 ALOGW("sampleRate() unknown thread %d", output); 539 return 0; 540 } 541 return thread->sampleRate(); 542} 543 544int AudioFlinger::channelCount(audio_io_handle_t output) const 545{ 546 Mutex::Autolock _l(mLock); 547 PlaybackThread *thread = checkPlaybackThread_l(output); 548 if (thread == NULL) { 549 ALOGW("channelCount() unknown thread %d", output); 550 return 0; 551 } 552 return thread->channelCount(); 553} 554 555audio_format_t AudioFlinger::format(audio_io_handle_t output) const 556{ 557 Mutex::Autolock _l(mLock); 558 PlaybackThread *thread = checkPlaybackThread_l(output); 559 if (thread == NULL) { 560 ALOGW("format() unknown thread %d", output); 561 return AUDIO_FORMAT_INVALID; 562 } 563 return thread->format(); 564} 565 566size_t AudioFlinger::frameCount(audio_io_handle_t output) const 567{ 568 Mutex::Autolock _l(mLock); 569 PlaybackThread *thread = checkPlaybackThread_l(output); 570 if (thread == NULL) { 571 ALOGW("frameCount() unknown thread %d", output); 572 return 0; 573 } 574 return thread->frameCount(); 575} 576 577uint32_t AudioFlinger::latency(audio_io_handle_t output) const 578{ 579 Mutex::Autolock _l(mLock); 580 PlaybackThread *thread = checkPlaybackThread_l(output); 581 if (thread == NULL) { 582 ALOGW("latency() unknown thread %d", output); 583 return 0; 584 } 585 return thread->latency(); 586} 587 588status_t AudioFlinger::setMasterVolume(float value) 589{ 590 status_t ret = initCheck(); 591 if (ret != NO_ERROR) { 592 return ret; 593 } 594 595 // check calling permissions 596 if (!settingsAllowed()) { 597 return PERMISSION_DENIED; 598 } 599 600 float swmv = value; 601 602 // when hw supports master volume, don't scale in sw mixer 603 if (MVS_NONE != mMasterVolumeSupportLvl) { 604 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 605 AutoMutex lock(mHardwareLock); 606 audio_hw_device_t *dev = mAudioHwDevs[i]; 607 608 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 609 if (NULL != dev->set_master_volume) { 610 dev->set_master_volume(dev, value); 611 } 612 mHardwareStatus = AUDIO_HW_IDLE; 613 } 614 615 swmv = 1.0; 616 } 617 618 Mutex::Autolock _l(mLock); 619 mMasterVolume = value; 620 mMasterVolumeSW = swmv; 621 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 622 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 623 624 return NO_ERROR; 625} 626 627status_t AudioFlinger::setMode(audio_mode_t mode) 628{ 629 status_t ret = initCheck(); 630 if (ret != NO_ERROR) { 631 return ret; 632 } 633 634 // check calling permissions 635 if (!settingsAllowed()) { 636 return PERMISSION_DENIED; 637 } 638 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 639 ALOGW("Illegal value: setMode(%d)", mode); 640 return BAD_VALUE; 641 } 642 643 { // scope for the lock 644 AutoMutex lock(mHardwareLock); 645 mHardwareStatus = AUDIO_HW_SET_MODE; 646 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 647 mHardwareStatus = AUDIO_HW_IDLE; 648 } 649 650 if (NO_ERROR == ret) { 651 Mutex::Autolock _l(mLock); 652 mMode = mode; 653 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 654 mPlaybackThreads.valueAt(i)->setMode(mode); 655 } 656 657 return ret; 658} 659 660status_t AudioFlinger::setMicMute(bool state) 661{ 662 status_t ret = initCheck(); 663 if (ret != NO_ERROR) { 664 return ret; 665 } 666 667 // check calling permissions 668 if (!settingsAllowed()) { 669 return PERMISSION_DENIED; 670 } 671 672 AutoMutex lock(mHardwareLock); 673 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 674 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 675 mHardwareStatus = AUDIO_HW_IDLE; 676 return ret; 677} 678 679bool AudioFlinger::getMicMute() const 680{ 681 status_t ret = initCheck(); 682 if (ret != NO_ERROR) { 683 return false; 684 } 685 686 bool state = AUDIO_MODE_INVALID; 687 AutoMutex lock(mHardwareLock); 688 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 689 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 690 mHardwareStatus = AUDIO_HW_IDLE; 691 return state; 692} 693 694status_t AudioFlinger::setMasterMute(bool muted) 695{ 696 // check calling permissions 697 if (!settingsAllowed()) { 698 return PERMISSION_DENIED; 699 } 700 701 Mutex::Autolock _l(mLock); 702 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 703 mMasterMute = muted; 704 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 705 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 706 707 return NO_ERROR; 708} 709 710float AudioFlinger::masterVolume() const 711{ 712 Mutex::Autolock _l(mLock); 713 return masterVolume_l(); 714} 715 716float AudioFlinger::masterVolumeSW() const 717{ 718 Mutex::Autolock _l(mLock); 719 return masterVolumeSW_l(); 720} 721 722bool AudioFlinger::masterMute() const 723{ 724 Mutex::Autolock _l(mLock); 725 return masterMute_l(); 726} 727 728float AudioFlinger::masterVolume_l() const 729{ 730 if (MVS_FULL == mMasterVolumeSupportLvl) { 731 float ret_val; 732 AutoMutex lock(mHardwareLock); 733 734 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 735 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 736 (NULL != mPrimaryHardwareDev->get_master_volume), 737 "can't get master volume"); 738 739 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 740 mHardwareStatus = AUDIO_HW_IDLE; 741 return ret_val; 742 } 743 744 return mMasterVolume; 745} 746 747status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 748 audio_io_handle_t output) 749{ 750 // check calling permissions 751 if (!settingsAllowed()) { 752 return PERMISSION_DENIED; 753 } 754 755 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 756 ALOGE("setStreamVolume() invalid stream %d", stream); 757 return BAD_VALUE; 758 } 759 760 AutoMutex lock(mLock); 761 PlaybackThread *thread = NULL; 762 if (output) { 763 thread = checkPlaybackThread_l(output); 764 if (thread == NULL) { 765 return BAD_VALUE; 766 } 767 } 768 769 mStreamTypes[stream].volume = value; 770 771 if (thread == NULL) { 772 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 773 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 774 } 775 } else { 776 thread->setStreamVolume(stream, value); 777 } 778 779 return NO_ERROR; 780} 781 782status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 783{ 784 // check calling permissions 785 if (!settingsAllowed()) { 786 return PERMISSION_DENIED; 787 } 788 789 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 790 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 791 ALOGE("setStreamMute() invalid stream %d", stream); 792 return BAD_VALUE; 793 } 794 795 AutoMutex lock(mLock); 796 mStreamTypes[stream].mute = muted; 797 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 798 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 799 800 return NO_ERROR; 801} 802 803float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 804{ 805 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 806 return 0.0f; 807 } 808 809 AutoMutex lock(mLock); 810 float volume; 811 if (output) { 812 PlaybackThread *thread = checkPlaybackThread_l(output); 813 if (thread == NULL) { 814 return 0.0f; 815 } 816 volume = thread->streamVolume(stream); 817 } else { 818 volume = streamVolume_l(stream); 819 } 820 821 return volume; 822} 823 824bool AudioFlinger::streamMute(audio_stream_type_t stream) const 825{ 826 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 827 return true; 828 } 829 830 AutoMutex lock(mLock); 831 return streamMute_l(stream); 832} 833 834status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 835{ 836 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 837 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 838 // check calling permissions 839 if (!settingsAllowed()) { 840 return PERMISSION_DENIED; 841 } 842 843 // ioHandle == 0 means the parameters are global to the audio hardware interface 844 if (ioHandle == 0) { 845 status_t final_result = NO_ERROR; 846 { 847 AutoMutex lock(mHardwareLock); 848 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 849 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 850 audio_hw_device_t *dev = mAudioHwDevs[i]; 851 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 852 final_result = result ?: final_result; 853 } 854 mHardwareStatus = AUDIO_HW_IDLE; 855 } 856 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 857 AudioParameter param = AudioParameter(keyValuePairs); 858 String8 value; 859 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 860 Mutex::Autolock _l(mLock); 861 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 862 if (mBtNrecIsOff != btNrecIsOff) { 863 for (size_t i = 0; i < mRecordThreads.size(); i++) { 864 sp<RecordThread> thread = mRecordThreads.valueAt(i); 865 RecordThread::RecordTrack *track = thread->track(); 866 if (track != NULL) { 867 audio_devices_t device = (audio_devices_t)( 868 thread->device() & AUDIO_DEVICE_IN_ALL); 869 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 870 thread->setEffectSuspended(FX_IID_AEC, 871 suspend, 872 track->sessionId()); 873 thread->setEffectSuspended(FX_IID_NS, 874 suspend, 875 track->sessionId()); 876 } 877 } 878 mBtNrecIsOff = btNrecIsOff; 879 } 880 } 881 return final_result; 882 } 883 884 // hold a strong ref on thread in case closeOutput() or closeInput() is called 885 // and the thread is exited once the lock is released 886 sp<ThreadBase> thread; 887 { 888 Mutex::Autolock _l(mLock); 889 thread = checkPlaybackThread_l(ioHandle); 890 if (thread == NULL) { 891 thread = checkRecordThread_l(ioHandle); 892 } else if (thread == primaryPlaybackThread_l()) { 893 // indicate output device change to all input threads for pre processing 894 AudioParameter param = AudioParameter(keyValuePairs); 895 int value; 896 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 897 (value != 0)) { 898 for (size_t i = 0; i < mRecordThreads.size(); i++) { 899 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 900 } 901 } 902 } 903 } 904 if (thread != 0) { 905 return thread->setParameters(keyValuePairs); 906 } 907 return BAD_VALUE; 908} 909 910String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 911{ 912// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 913// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 914 915 if (ioHandle == 0) { 916 String8 out_s8; 917 918 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 919 char *s; 920 { 921 AutoMutex lock(mHardwareLock); 922 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 923 audio_hw_device_t *dev = mAudioHwDevs[i]; 924 s = dev->get_parameters(dev, keys.string()); 925 mHardwareStatus = AUDIO_HW_IDLE; 926 } 927 out_s8 += String8(s ? s : ""); 928 free(s); 929 } 930 return out_s8; 931 } 932 933 Mutex::Autolock _l(mLock); 934 935 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 936 if (playbackThread != NULL) { 937 return playbackThread->getParameters(keys); 938 } 939 RecordThread *recordThread = checkRecordThread_l(ioHandle); 940 if (recordThread != NULL) { 941 return recordThread->getParameters(keys); 942 } 943 return String8(""); 944} 945 946size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 947{ 948 status_t ret = initCheck(); 949 if (ret != NO_ERROR) { 950 return 0; 951 } 952 953 AutoMutex lock(mHardwareLock); 954 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 955 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 956 mHardwareStatus = AUDIO_HW_IDLE; 957 return size; 958} 959 960unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 961{ 962 if (ioHandle == 0) { 963 return 0; 964 } 965 966 Mutex::Autolock _l(mLock); 967 968 RecordThread *recordThread = checkRecordThread_l(ioHandle); 969 if (recordThread != NULL) { 970 return recordThread->getInputFramesLost(); 971 } 972 return 0; 973} 974 975status_t AudioFlinger::setVoiceVolume(float value) 976{ 977 status_t ret = initCheck(); 978 if (ret != NO_ERROR) { 979 return ret; 980 } 981 982 // check calling permissions 983 if (!settingsAllowed()) { 984 return PERMISSION_DENIED; 985 } 986 987 AutoMutex lock(mHardwareLock); 988 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 989 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 990 mHardwareStatus = AUDIO_HW_IDLE; 991 992 return ret; 993} 994 995status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 996 audio_io_handle_t output) const 997{ 998 status_t status; 999 1000 Mutex::Autolock _l(mLock); 1001 1002 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1003 if (playbackThread != NULL) { 1004 return playbackThread->getRenderPosition(halFrames, dspFrames); 1005 } 1006 1007 return BAD_VALUE; 1008} 1009 1010void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1011{ 1012 1013 Mutex::Autolock _l(mLock); 1014 1015 pid_t pid = IPCThreadState::self()->getCallingPid(); 1016 if (mNotificationClients.indexOfKey(pid) < 0) { 1017 sp<NotificationClient> notificationClient = new NotificationClient(this, 1018 client, 1019 pid); 1020 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1021 1022 mNotificationClients.add(pid, notificationClient); 1023 1024 sp<IBinder> binder = client->asBinder(); 1025 binder->linkToDeath(notificationClient); 1026 1027 // the config change is always sent from playback or record threads to avoid deadlock 1028 // with AudioSystem::gLock 1029 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1030 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1031 } 1032 1033 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1034 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1035 } 1036 } 1037} 1038 1039void AudioFlinger::removeNotificationClient(pid_t pid) 1040{ 1041 Mutex::Autolock _l(mLock); 1042 1043 mNotificationClients.removeItem(pid); 1044 1045 ALOGV("%d died, releasing its sessions", pid); 1046 size_t num = mAudioSessionRefs.size(); 1047 bool removed = false; 1048 for (size_t i = 0; i< num; ) { 1049 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1050 ALOGV(" pid %d @ %d", ref->mPid, i); 1051 if (ref->mPid == pid) { 1052 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1053 mAudioSessionRefs.removeAt(i); 1054 delete ref; 1055 removed = true; 1056 num--; 1057 } else { 1058 i++; 1059 } 1060 } 1061 if (removed) { 1062 purgeStaleEffects_l(); 1063 } 1064} 1065 1066// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1067void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1068{ 1069 size_t size = mNotificationClients.size(); 1070 for (size_t i = 0; i < size; i++) { 1071 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1072 param2); 1073 } 1074} 1075 1076// removeClient_l() must be called with AudioFlinger::mLock held 1077void AudioFlinger::removeClient_l(pid_t pid) 1078{ 1079 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1080 mClients.removeItem(pid); 1081} 1082 1083 1084// ---------------------------------------------------------------------------- 1085 1086AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1087 uint32_t device, type_t type) 1088 : Thread(false), 1089 mType(type), 1090 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1091 // mChannelMask 1092 mChannelCount(0), 1093 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1094 mParamStatus(NO_ERROR), 1095 mStandby(false), mId(id), 1096 mDevice(device), 1097 mDeathRecipient(new PMDeathRecipient(this)) 1098{ 1099} 1100 1101AudioFlinger::ThreadBase::~ThreadBase() 1102{ 1103 mParamCond.broadcast(); 1104 // do not lock the mutex in destructor 1105 releaseWakeLock_l(); 1106 if (mPowerManager != 0) { 1107 sp<IBinder> binder = mPowerManager->asBinder(); 1108 binder->unlinkToDeath(mDeathRecipient); 1109 } 1110} 1111 1112void AudioFlinger::ThreadBase::exit() 1113{ 1114 ALOGV("ThreadBase::exit"); 1115 { 1116 // This lock prevents the following race in thread (uniprocessor for illustration): 1117 // if (!exitPending()) { 1118 // // context switch from here to exit() 1119 // // exit() calls requestExit(), what exitPending() observes 1120 // // exit() calls signal(), which is dropped since no waiters 1121 // // context switch back from exit() to here 1122 // mWaitWorkCV.wait(...); 1123 // // now thread is hung 1124 // } 1125 AutoMutex lock(mLock); 1126 requestExit(); 1127 mWaitWorkCV.signal(); 1128 } 1129 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1130 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1131 requestExitAndWait(); 1132} 1133 1134status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1135{ 1136 status_t status; 1137 1138 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1139 Mutex::Autolock _l(mLock); 1140 1141 mNewParameters.add(keyValuePairs); 1142 mWaitWorkCV.signal(); 1143 // wait condition with timeout in case the thread loop has exited 1144 // before the request could be processed 1145 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1146 status = mParamStatus; 1147 mWaitWorkCV.signal(); 1148 } else { 1149 status = TIMED_OUT; 1150 } 1151 return status; 1152} 1153 1154void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1155{ 1156 Mutex::Autolock _l(mLock); 1157 sendConfigEvent_l(event, param); 1158} 1159 1160// sendConfigEvent_l() must be called with ThreadBase::mLock held 1161void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1162{ 1163 ConfigEvent configEvent; 1164 configEvent.mEvent = event; 1165 configEvent.mParam = param; 1166 mConfigEvents.add(configEvent); 1167 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1168 mWaitWorkCV.signal(); 1169} 1170 1171void AudioFlinger::ThreadBase::processConfigEvents() 1172{ 1173 mLock.lock(); 1174 while (!mConfigEvents.isEmpty()) { 1175 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1176 ConfigEvent configEvent = mConfigEvents[0]; 1177 mConfigEvents.removeAt(0); 1178 // release mLock before locking AudioFlinger mLock: lock order is always 1179 // AudioFlinger then ThreadBase to avoid cross deadlock 1180 mLock.unlock(); 1181 mAudioFlinger->mLock.lock(); 1182 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1183 mAudioFlinger->mLock.unlock(); 1184 mLock.lock(); 1185 } 1186 mLock.unlock(); 1187} 1188 1189status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1190{ 1191 const size_t SIZE = 256; 1192 char buffer[SIZE]; 1193 String8 result; 1194 1195 bool locked = tryLock(mLock); 1196 if (!locked) { 1197 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1198 write(fd, buffer, strlen(buffer)); 1199 } 1200 1201 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1202 result.append(buffer); 1203 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1204 result.append(buffer); 1205 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1206 result.append(buffer); 1207 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1208 result.append(buffer); 1209 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1210 result.append(buffer); 1211 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1212 result.append(buffer); 1213 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1214 result.append(buffer); 1215 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1216 result.append(buffer); 1217 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1218 result.append(buffer); 1219 1220 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1221 result.append(buffer); 1222 result.append(" Index Command"); 1223 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1224 snprintf(buffer, SIZE, "\n %02d ", i); 1225 result.append(buffer); 1226 result.append(mNewParameters[i]); 1227 } 1228 1229 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, " Index event param\n"); 1232 result.append(buffer); 1233 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1234 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1235 result.append(buffer); 1236 } 1237 result.append("\n"); 1238 1239 write(fd, result.string(), result.size()); 1240 1241 if (locked) { 1242 mLock.unlock(); 1243 } 1244 return NO_ERROR; 1245} 1246 1247status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1248{ 1249 const size_t SIZE = 256; 1250 char buffer[SIZE]; 1251 String8 result; 1252 1253 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1254 write(fd, buffer, strlen(buffer)); 1255 1256 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1257 sp<EffectChain> chain = mEffectChains[i]; 1258 if (chain != 0) { 1259 chain->dump(fd, args); 1260 } 1261 } 1262 return NO_ERROR; 1263} 1264 1265void AudioFlinger::ThreadBase::acquireWakeLock() 1266{ 1267 Mutex::Autolock _l(mLock); 1268 acquireWakeLock_l(); 1269} 1270 1271void AudioFlinger::ThreadBase::acquireWakeLock_l() 1272{ 1273 if (mPowerManager == 0) { 1274 // use checkService() to avoid blocking if power service is not up yet 1275 sp<IBinder> binder = 1276 defaultServiceManager()->checkService(String16("power")); 1277 if (binder == 0) { 1278 ALOGW("Thread %s cannot connect to the power manager service", mName); 1279 } else { 1280 mPowerManager = interface_cast<IPowerManager>(binder); 1281 binder->linkToDeath(mDeathRecipient); 1282 } 1283 } 1284 if (mPowerManager != 0) { 1285 sp<IBinder> binder = new BBinder(); 1286 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1287 binder, 1288 String16(mName)); 1289 if (status == NO_ERROR) { 1290 mWakeLockToken = binder; 1291 } 1292 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1293 } 1294} 1295 1296void AudioFlinger::ThreadBase::releaseWakeLock() 1297{ 1298 Mutex::Autolock _l(mLock); 1299 releaseWakeLock_l(); 1300} 1301 1302void AudioFlinger::ThreadBase::releaseWakeLock_l() 1303{ 1304 if (mWakeLockToken != 0) { 1305 ALOGV("releaseWakeLock_l() %s", mName); 1306 if (mPowerManager != 0) { 1307 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1308 } 1309 mWakeLockToken.clear(); 1310 } 1311} 1312 1313void AudioFlinger::ThreadBase::clearPowerManager() 1314{ 1315 Mutex::Autolock _l(mLock); 1316 releaseWakeLock_l(); 1317 mPowerManager.clear(); 1318} 1319 1320void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1321{ 1322 sp<ThreadBase> thread = mThread.promote(); 1323 if (thread != 0) { 1324 thread->clearPowerManager(); 1325 } 1326 ALOGW("power manager service died !!!"); 1327} 1328 1329void AudioFlinger::ThreadBase::setEffectSuspended( 1330 const effect_uuid_t *type, bool suspend, int sessionId) 1331{ 1332 Mutex::Autolock _l(mLock); 1333 setEffectSuspended_l(type, suspend, sessionId); 1334} 1335 1336void AudioFlinger::ThreadBase::setEffectSuspended_l( 1337 const effect_uuid_t *type, bool suspend, int sessionId) 1338{ 1339 sp<EffectChain> chain = getEffectChain_l(sessionId); 1340 if (chain != 0) { 1341 if (type != NULL) { 1342 chain->setEffectSuspended_l(type, suspend); 1343 } else { 1344 chain->setEffectSuspendedAll_l(suspend); 1345 } 1346 } 1347 1348 updateSuspendedSessions_l(type, suspend, sessionId); 1349} 1350 1351void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1352{ 1353 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1354 if (index < 0) { 1355 return; 1356 } 1357 1358 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1359 mSuspendedSessions.editValueAt(index); 1360 1361 for (size_t i = 0; i < sessionEffects.size(); i++) { 1362 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1363 for (int j = 0; j < desc->mRefCount; j++) { 1364 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1365 chain->setEffectSuspendedAll_l(true); 1366 } else { 1367 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1368 desc->mType.timeLow); 1369 chain->setEffectSuspended_l(&desc->mType, true); 1370 } 1371 } 1372 } 1373} 1374 1375void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1376 bool suspend, 1377 int sessionId) 1378{ 1379 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1380 1381 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1382 1383 if (suspend) { 1384 if (index >= 0) { 1385 sessionEffects = mSuspendedSessions.editValueAt(index); 1386 } else { 1387 mSuspendedSessions.add(sessionId, sessionEffects); 1388 } 1389 } else { 1390 if (index < 0) { 1391 return; 1392 } 1393 sessionEffects = mSuspendedSessions.editValueAt(index); 1394 } 1395 1396 1397 int key = EffectChain::kKeyForSuspendAll; 1398 if (type != NULL) { 1399 key = type->timeLow; 1400 } 1401 index = sessionEffects.indexOfKey(key); 1402 1403 sp<SuspendedSessionDesc> desc; 1404 if (suspend) { 1405 if (index >= 0) { 1406 desc = sessionEffects.valueAt(index); 1407 } else { 1408 desc = new SuspendedSessionDesc(); 1409 if (type != NULL) { 1410 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1411 } 1412 sessionEffects.add(key, desc); 1413 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1414 } 1415 desc->mRefCount++; 1416 } else { 1417 if (index < 0) { 1418 return; 1419 } 1420 desc = sessionEffects.valueAt(index); 1421 if (--desc->mRefCount == 0) { 1422 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1423 sessionEffects.removeItemsAt(index); 1424 if (sessionEffects.isEmpty()) { 1425 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1426 sessionId); 1427 mSuspendedSessions.removeItem(sessionId); 1428 } 1429 } 1430 } 1431 if (!sessionEffects.isEmpty()) { 1432 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1433 } 1434} 1435 1436void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1437 bool enabled, 1438 int sessionId) 1439{ 1440 Mutex::Autolock _l(mLock); 1441 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1442} 1443 1444void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1445 bool enabled, 1446 int sessionId) 1447{ 1448 if (mType != RECORD) { 1449 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1450 // another session. This gives the priority to well behaved effect control panels 1451 // and applications not using global effects. 1452 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1453 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1454 } 1455 } 1456 1457 sp<EffectChain> chain = getEffectChain_l(sessionId); 1458 if (chain != 0) { 1459 chain->checkSuspendOnEffectEnabled(effect, enabled); 1460 } 1461} 1462 1463// ---------------------------------------------------------------------------- 1464 1465AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1466 AudioStreamOut* output, 1467 audio_io_handle_t id, 1468 uint32_t device, 1469 type_t type) 1470 : ThreadBase(audioFlinger, id, device, type), 1471 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1472 // Assumes constructor is called by AudioFlinger with it's mLock held, 1473 // but it would be safer to explicitly pass initial masterMute as parameter 1474 mMasterMute(audioFlinger->masterMute_l()), 1475 // mStreamTypes[] initialized in constructor body 1476 mOutput(output), 1477 // Assumes constructor is called by AudioFlinger with it's mLock held, 1478 // but it would be safer to explicitly pass initial masterVolume as parameter 1479 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1480 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1481 mMixerStatus(MIXER_IDLE), 1482 mPrevMixerStatus(MIXER_IDLE), 1483 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1484{ 1485 snprintf(mName, kNameLength, "AudioOut_%X", id); 1486 1487 readOutputParameters(); 1488 1489 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1490 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1491 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1492 stream = (audio_stream_type_t) (stream + 1)) { 1493 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1494 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1495 // initialized by stream_type_t default constructor 1496 // mStreamTypes[stream].valid = true; 1497 } 1498 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1499 // because mAudioFlinger doesn't have one to copy from 1500} 1501 1502AudioFlinger::PlaybackThread::~PlaybackThread() 1503{ 1504 delete [] mMixBuffer; 1505} 1506 1507status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1508{ 1509 dumpInternals(fd, args); 1510 dumpTracks(fd, args); 1511 dumpEffectChains(fd, args); 1512 return NO_ERROR; 1513} 1514 1515status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1516{ 1517 const size_t SIZE = 256; 1518 char buffer[SIZE]; 1519 String8 result; 1520 1521 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1522 result.append(buffer); 1523 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1524 for (size_t i = 0; i < mTracks.size(); ++i) { 1525 sp<Track> track = mTracks[i]; 1526 if (track != 0) { 1527 track->dump(buffer, SIZE); 1528 result.append(buffer); 1529 } 1530 } 1531 1532 snprintf(buffer, SIZE, "Output thread %p active 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 < mActiveTracks.size(); ++i) { 1536 sp<Track> track = mActiveTracks[i].promote(); 1537 if (track != 0) { 1538 track->dump(buffer, SIZE); 1539 result.append(buffer); 1540 } 1541 } 1542 write(fd, result.string(), result.size()); 1543 return NO_ERROR; 1544} 1545 1546status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1547{ 1548 const size_t SIZE = 256; 1549 char buffer[SIZE]; 1550 String8 result; 1551 1552 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1553 result.append(buffer); 1554 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1555 result.append(buffer); 1556 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1557 result.append(buffer); 1558 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1559 result.append(buffer); 1560 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1561 result.append(buffer); 1562 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1563 result.append(buffer); 1564 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1565 result.append(buffer); 1566 write(fd, result.string(), result.size()); 1567 1568 dumpBase(fd, args); 1569 1570 return NO_ERROR; 1571} 1572 1573// Thread virtuals 1574status_t AudioFlinger::PlaybackThread::readyToRun() 1575{ 1576 status_t status = initCheck(); 1577 if (status == NO_ERROR) { 1578 ALOGI("AudioFlinger's thread %p ready to run", this); 1579 } else { 1580 ALOGE("No working audio driver found."); 1581 } 1582 return status; 1583} 1584 1585void AudioFlinger::PlaybackThread::onFirstRef() 1586{ 1587 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1588} 1589 1590// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1591sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1592 const sp<AudioFlinger::Client>& client, 1593 audio_stream_type_t streamType, 1594 uint32_t sampleRate, 1595 audio_format_t format, 1596 uint32_t channelMask, 1597 int frameCount, 1598 const sp<IMemory>& sharedBuffer, 1599 int sessionId, 1600 bool isTimed, 1601 status_t *status) 1602{ 1603 sp<Track> track; 1604 status_t lStatus; 1605 1606 if (mType == DIRECT) { 1607 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1608 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1609 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1610 "for output %p with format %d", 1611 sampleRate, format, channelMask, mOutput, mFormat); 1612 lStatus = BAD_VALUE; 1613 goto Exit; 1614 } 1615 } 1616 } else { 1617 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1618 if (sampleRate > mSampleRate*2) { 1619 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1620 lStatus = BAD_VALUE; 1621 goto Exit; 1622 } 1623 } 1624 1625 lStatus = initCheck(); 1626 if (lStatus != NO_ERROR) { 1627 ALOGE("Audio driver not initialized."); 1628 goto Exit; 1629 } 1630 1631 { // scope for mLock 1632 Mutex::Autolock _l(mLock); 1633 1634 // all tracks in same audio session must share the same routing strategy otherwise 1635 // conflicts will happen when tracks are moved from one output to another by audio policy 1636 // manager 1637 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1638 for (size_t i = 0; i < mTracks.size(); ++i) { 1639 sp<Track> t = mTracks[i]; 1640 if (t != 0 && !t->isOutputTrack()) { 1641 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1642 if (sessionId == t->sessionId() && strategy != actual) { 1643 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1644 strategy, actual); 1645 lStatus = BAD_VALUE; 1646 goto Exit; 1647 } 1648 } 1649 } 1650 1651 if (!isTimed) { 1652 track = new Track(this, client, streamType, sampleRate, format, 1653 channelMask, frameCount, sharedBuffer, sessionId); 1654 } else { 1655 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1656 channelMask, frameCount, sharedBuffer, sessionId); 1657 } 1658 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1659 lStatus = NO_MEMORY; 1660 goto Exit; 1661 } 1662 mTracks.add(track); 1663 1664 sp<EffectChain> chain = getEffectChain_l(sessionId); 1665 if (chain != 0) { 1666 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1667 track->setMainBuffer(chain->inBuffer()); 1668 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1669 chain->incTrackCnt(); 1670 } 1671 1672 // invalidate track immediately if the stream type was moved to another thread since 1673 // createTrack() was called by the client process. 1674 if (!mStreamTypes[streamType].valid) { 1675 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1676 this, streamType); 1677 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1678 } 1679 } 1680 lStatus = NO_ERROR; 1681 1682Exit: 1683 if (status) { 1684 *status = lStatus; 1685 } 1686 return track; 1687} 1688 1689uint32_t AudioFlinger::PlaybackThread::latency() const 1690{ 1691 Mutex::Autolock _l(mLock); 1692 if (initCheck() == NO_ERROR) { 1693 return mOutput->stream->get_latency(mOutput->stream); 1694 } else { 1695 return 0; 1696 } 1697} 1698 1699void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1700{ 1701 Mutex::Autolock _l(mLock); 1702 mMasterVolume = value; 1703} 1704 1705void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1706{ 1707 Mutex::Autolock _l(mLock); 1708 setMasterMute_l(muted); 1709} 1710 1711void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1712{ 1713 Mutex::Autolock _l(mLock); 1714 mStreamTypes[stream].volume = value; 1715} 1716 1717void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1718{ 1719 Mutex::Autolock _l(mLock); 1720 mStreamTypes[stream].mute = muted; 1721} 1722 1723float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1724{ 1725 Mutex::Autolock _l(mLock); 1726 return mStreamTypes[stream].volume; 1727} 1728 1729// addTrack_l() must be called with ThreadBase::mLock held 1730status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1731{ 1732 status_t status = ALREADY_EXISTS; 1733 1734 // set retry count for buffer fill 1735 track->mRetryCount = kMaxTrackStartupRetries; 1736 if (mActiveTracks.indexOf(track) < 0) { 1737 // the track is newly added, make sure it fills up all its 1738 // buffers before playing. This is to ensure the client will 1739 // effectively get the latency it requested. 1740 track->mFillingUpStatus = Track::FS_FILLING; 1741 track->mResetDone = false; 1742 mActiveTracks.add(track); 1743 if (track->mainBuffer() != mMixBuffer) { 1744 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1745 if (chain != 0) { 1746 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1747 chain->incActiveTrackCnt(); 1748 } 1749 } 1750 1751 status = NO_ERROR; 1752 } 1753 1754 ALOGV("mWaitWorkCV.broadcast"); 1755 mWaitWorkCV.broadcast(); 1756 1757 return status; 1758} 1759 1760// destroyTrack_l() must be called with ThreadBase::mLock held 1761void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1762{ 1763 track->mState = TrackBase::TERMINATED; 1764 if (mActiveTracks.indexOf(track) < 0) { 1765 removeTrack_l(track); 1766 } 1767} 1768 1769void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1770{ 1771 mTracks.remove(track); 1772 deleteTrackName_l(track->name()); 1773 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1774 if (chain != 0) { 1775 chain->decTrackCnt(); 1776 } 1777} 1778 1779String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1780{ 1781 String8 out_s8 = String8(""); 1782 char *s; 1783 1784 Mutex::Autolock _l(mLock); 1785 if (initCheck() != NO_ERROR) { 1786 return out_s8; 1787 } 1788 1789 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1790 out_s8 = String8(s); 1791 free(s); 1792 return out_s8; 1793} 1794 1795// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1796void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1797 AudioSystem::OutputDescriptor desc; 1798 void *param2 = NULL; 1799 1800 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1801 1802 switch (event) { 1803 case AudioSystem::OUTPUT_OPENED: 1804 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1805 desc.channels = mChannelMask; 1806 desc.samplingRate = mSampleRate; 1807 desc.format = mFormat; 1808 desc.frameCount = mFrameCount; 1809 desc.latency = latency(); 1810 param2 = &desc; 1811 break; 1812 1813 case AudioSystem::STREAM_CONFIG_CHANGED: 1814 param2 = ¶m; 1815 case AudioSystem::OUTPUT_CLOSED: 1816 default: 1817 break; 1818 } 1819 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1820} 1821 1822void AudioFlinger::PlaybackThread::readOutputParameters() 1823{ 1824 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1825 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1826 mChannelCount = (uint16_t)popcount(mChannelMask); 1827 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1828 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1829 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1830 1831 // FIXME - Current mixer implementation only supports stereo output: Always 1832 // Allocate a stereo buffer even if HW output is mono. 1833 delete[] mMixBuffer; 1834 mMixBuffer = new int16_t[mFrameCount * 2]; 1835 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1836 1837 // force reconfiguration of effect chains and engines to take new buffer size and audio 1838 // parameters into account 1839 // Note that mLock is not held when readOutputParameters() is called from the constructor 1840 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1841 // matter. 1842 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1843 Vector< sp<EffectChain> > effectChains = mEffectChains; 1844 for (size_t i = 0; i < effectChains.size(); i ++) { 1845 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1846 } 1847} 1848 1849status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1850{ 1851 if (halFrames == NULL || dspFrames == NULL) { 1852 return BAD_VALUE; 1853 } 1854 Mutex::Autolock _l(mLock); 1855 if (initCheck() != NO_ERROR) { 1856 return INVALID_OPERATION; 1857 } 1858 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1859 1860 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1861} 1862 1863uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1864{ 1865 Mutex::Autolock _l(mLock); 1866 uint32_t result = 0; 1867 if (getEffectChain_l(sessionId) != 0) { 1868 result = EFFECT_SESSION; 1869 } 1870 1871 for (size_t i = 0; i < mTracks.size(); ++i) { 1872 sp<Track> track = mTracks[i]; 1873 if (sessionId == track->sessionId() && 1874 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1875 result |= TRACK_SESSION; 1876 break; 1877 } 1878 } 1879 1880 return result; 1881} 1882 1883uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1884{ 1885 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1886 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1887 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1888 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1889 } 1890 for (size_t i = 0; i < mTracks.size(); i++) { 1891 sp<Track> track = mTracks[i]; 1892 if (sessionId == track->sessionId() && 1893 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1894 return AudioSystem::getStrategyForStream(track->streamType()); 1895 } 1896 } 1897 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1898} 1899 1900 1901AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1902{ 1903 Mutex::Autolock _l(mLock); 1904 return mOutput; 1905} 1906 1907AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1908{ 1909 Mutex::Autolock _l(mLock); 1910 AudioStreamOut *output = mOutput; 1911 mOutput = NULL; 1912 return output; 1913} 1914 1915// this method must always be called either with ThreadBase mLock held or inside the thread loop 1916audio_stream_t* AudioFlinger::PlaybackThread::stream() 1917{ 1918 if (mOutput == NULL) { 1919 return NULL; 1920 } 1921 return &mOutput->stream->common; 1922} 1923 1924uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1925{ 1926 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1927 // decoding and transfer time. So sleeping for half of the latency would likely cause 1928 // underruns 1929 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1930 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1931 } else { 1932 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1933 } 1934} 1935 1936// ---------------------------------------------------------------------------- 1937 1938AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1939 audio_io_handle_t id, uint32_t device, type_t type) 1940 : PlaybackThread(audioFlinger, output, id, device, type) 1941{ 1942 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1943 // FIXME - Current mixer implementation only supports stereo output 1944 if (mChannelCount == 1) { 1945 ALOGE("Invalid audio hardware channel count"); 1946 } 1947} 1948 1949AudioFlinger::MixerThread::~MixerThread() 1950{ 1951 delete mAudioMixer; 1952} 1953 1954class CpuStats { 1955public: 1956 CpuStats(); 1957 void sample(const String8 &title); 1958#ifdef DEBUG_CPU_USAGE 1959private: 1960 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 1961 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 1962 1963 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 1964 1965 int mCpuNum; // thread's current CPU number 1966 int mCpukHz; // frequency of thread's current CPU in kHz 1967#endif 1968}; 1969 1970CpuStats::CpuStats() 1971#ifdef DEBUG_CPU_USAGE 1972 : mCpuNum(-1), mCpukHz(-1) 1973#endif 1974{ 1975} 1976 1977void CpuStats::sample(const String8 &title) { 1978#ifdef DEBUG_CPU_USAGE 1979 // get current thread's delta CPU time in wall clock ns 1980 double wcNs; 1981 bool valid = mCpuUsage.sampleAndEnable(wcNs); 1982 1983 // record sample for wall clock statistics 1984 if (valid) { 1985 mWcStats.sample(wcNs); 1986 } 1987 1988 // get the current CPU number 1989 int cpuNum = sched_getcpu(); 1990 1991 // get the current CPU frequency in kHz 1992 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 1993 1994 // check if either CPU number or frequency changed 1995 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 1996 mCpuNum = cpuNum; 1997 mCpukHz = cpukHz; 1998 // ignore sample for purposes of cycles 1999 valid = false; 2000 } 2001 2002 // if no change in CPU number or frequency, then record sample for cycle statistics 2003 if (valid && mCpukHz > 0) { 2004 double cycles = wcNs * cpukHz * 0.000001; 2005 mHzStats.sample(cycles); 2006 } 2007 2008 unsigned n = mWcStats.n(); 2009 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2010 if ((n & 127) == 1) { 2011 long long elapsed = mCpuUsage.elapsed(); 2012 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2013 double perLoop = elapsed / (double) n; 2014 double perLoop100 = perLoop * 0.01; 2015 double perLoop1k = perLoop * 0.001; 2016 double mean = mWcStats.mean(); 2017 double stddev = mWcStats.stddev(); 2018 double minimum = mWcStats.minimum(); 2019 double maximum = mWcStats.maximum(); 2020 double meanCycles = mHzStats.mean(); 2021 double stddevCycles = mHzStats.stddev(); 2022 double minCycles = mHzStats.minimum(); 2023 double maxCycles = mHzStats.maximum(); 2024 mCpuUsage.resetElapsed(); 2025 mWcStats.reset(); 2026 mHzStats.reset(); 2027 ALOGD("CPU usage for %s over past %.1f secs\n" 2028 " (%u mixer loops at %.1f mean ms per loop):\n" 2029 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2030 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2031 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2032 title.string(), 2033 elapsed * .000000001, n, perLoop * .000001, 2034 mean * .001, 2035 stddev * .001, 2036 minimum * .001, 2037 maximum * .001, 2038 mean / perLoop100, 2039 stddev / perLoop100, 2040 minimum / perLoop100, 2041 maximum / perLoop100, 2042 meanCycles / perLoop1k, 2043 stddevCycles / perLoop1k, 2044 minCycles / perLoop1k, 2045 maxCycles / perLoop1k); 2046 2047 } 2048 } 2049#endif 2050}; 2051 2052void AudioFlinger::PlaybackThread::checkSilentMode_l() 2053{ 2054 if (!mMasterMute) { 2055 char value[PROPERTY_VALUE_MAX]; 2056 if (property_get("ro.audio.silent", value, "0") > 0) { 2057 char *endptr; 2058 unsigned long ul = strtoul(value, &endptr, 0); 2059 if (*endptr == '\0' && ul != 0) { 2060 ALOGD("Silence is golden"); 2061 // The setprop command will not allow a property to be changed after 2062 // the first time it is set, so we don't have to worry about un-muting. 2063 setMasterMute_l(true); 2064 } 2065 } 2066 } 2067} 2068 2069bool AudioFlinger::PlaybackThread::threadLoop() 2070{ 2071 Vector< sp<Track> > tracksToRemove; 2072 2073 standbyTime = systemTime(); 2074 2075 // MIXER 2076 nsecs_t lastWarning = 0; 2077if (mType == MIXER) { 2078 longStandbyExit = false; 2079} 2080 2081 // DUPLICATING 2082 // FIXME could this be made local to while loop? 2083 writeFrames = 0; 2084 2085 cacheParameters_l(); 2086 sleepTime = idleSleepTime; 2087 2088if (mType == MIXER) { 2089 sleepTimeShift = 0; 2090} 2091 2092 CpuStats cpuStats; 2093 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2094 2095 acquireWakeLock(); 2096 2097 while (!exitPending()) 2098 { 2099 cpuStats.sample(myName); 2100 2101 Vector< sp<EffectChain> > effectChains; 2102 2103 processConfigEvents(); 2104 2105 { // scope for mLock 2106 2107 Mutex::Autolock _l(mLock); 2108 2109 if (checkForNewParameters_l()) { 2110 cacheParameters_l(); 2111 } 2112 2113 saveOutputTracks(); 2114 2115 // put audio hardware into standby after short delay 2116 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2117 mSuspended > 0)) { 2118 if (!mStandby) { 2119 2120 threadLoop_standby(); 2121 2122 mStandby = true; 2123 mBytesWritten = 0; 2124 } 2125 2126 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2127 // we're about to wait, flush the binder command buffer 2128 IPCThreadState::self()->flushCommands(); 2129 2130 clearOutputTracks(); 2131 2132 if (exitPending()) break; 2133 2134 releaseWakeLock_l(); 2135 // wait until we have something to do... 2136 ALOGV("%s going to sleep", myName.string()); 2137 mWaitWorkCV.wait(mLock); 2138 ALOGV("%s waking up", myName.string()); 2139 acquireWakeLock_l(); 2140 2141 mPrevMixerStatus = MIXER_IDLE; 2142 2143 checkSilentMode_l(); 2144 2145 standbyTime = systemTime() + standbyDelay; 2146 sleepTime = idleSleepTime; 2147 if (mType == MIXER) { 2148 sleepTimeShift = 0; 2149 } 2150 2151 continue; 2152 } 2153 } 2154 2155 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2156 // Shift in the new status; this could be a queue if it's 2157 // useful to filter the mixer status over several cycles. 2158 mPrevMixerStatus = mMixerStatus; 2159 mMixerStatus = newMixerStatus; 2160 2161 // prevent any changes in effect chain list and in each effect chain 2162 // during mixing and effect process as the audio buffers could be deleted 2163 // or modified if an effect is created or deleted 2164 lockEffectChains_l(effectChains); 2165 } 2166 2167 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2168 threadLoop_mix(); 2169 } else { 2170 threadLoop_sleepTime(); 2171 } 2172 2173 if (mSuspended > 0) { 2174 sleepTime = suspendSleepTimeUs(); 2175 } 2176 2177 // only process effects if we're going to write 2178 if (sleepTime == 0) { 2179 for (size_t i = 0; i < effectChains.size(); i ++) { 2180 effectChains[i]->process_l(); 2181 } 2182 } 2183 2184 // enable changes in effect chain 2185 unlockEffectChains(effectChains); 2186 2187 // sleepTime == 0 means we must write to audio hardware 2188 if (sleepTime == 0) { 2189 2190 threadLoop_write(); 2191 2192if (mType == MIXER) { 2193 // write blocked detection 2194 nsecs_t now = systemTime(); 2195 nsecs_t delta = now - mLastWriteTime; 2196 if (!mStandby && delta > maxPeriod) { 2197 mNumDelayedWrites++; 2198 if ((now - lastWarning) > kWarningThrottleNs) { 2199 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2200 ns2ms(delta), mNumDelayedWrites, this); 2201 lastWarning = now; 2202 } 2203 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2204 // a different threshold. Or completely removed for what it is worth anyway... 2205 if (mStandby) { 2206 longStandbyExit = true; 2207 } 2208 } 2209} 2210 2211 mStandby = false; 2212 } else { 2213 usleep(sleepTime); 2214 } 2215 2216 // finally let go of removed track(s), without the lock held 2217 // since we can't guarantee the destructors won't acquire that 2218 // same lock. 2219 tracksToRemove.clear(); 2220 2221 // FIXME I don't understand the need for this here; 2222 // it was in the original code but maybe the 2223 // assignment in saveOutputTracks() makes this unnecessary? 2224 clearOutputTracks(); 2225 2226 // Effect chains will be actually deleted here if they were removed from 2227 // mEffectChains list during mixing or effects processing 2228 effectChains.clear(); 2229 2230 // FIXME Note that the above .clear() is no longer necessary since effectChains 2231 // is now local to this block, but will keep it for now (at least until merge done). 2232 } 2233 2234if (mType == MIXER || mType == DIRECT) { 2235 // put output stream into standby mode 2236 if (!mStandby) { 2237 mOutput->stream->common.standby(&mOutput->stream->common); 2238 } 2239} 2240if (mType == DUPLICATING) { 2241 // for DuplicatingThread, standby mode is handled by the outputTracks 2242} 2243 2244 releaseWakeLock(); 2245 2246 ALOGV("Thread %p type %d exiting", this, mType); 2247 return false; 2248} 2249 2250// shared by MIXER and DIRECT, overridden by DUPLICATING 2251void AudioFlinger::PlaybackThread::threadLoop_write() 2252{ 2253 // FIXME rewrite to reduce number of system calls 2254 mLastWriteTime = systemTime(); 2255 mInWrite = true; 2256 mBytesWritten += mixBufferSize; 2257 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2258 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2259 mNumWrites++; 2260 mInWrite = false; 2261} 2262 2263// shared by MIXER and DIRECT, overridden by DUPLICATING 2264void AudioFlinger::PlaybackThread::threadLoop_standby() 2265{ 2266 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2267 mOutput->stream->common.standby(&mOutput->stream->common); 2268} 2269 2270void AudioFlinger::MixerThread::threadLoop_mix() 2271{ 2272 // obtain the presentation timestamp of the next output buffer 2273 int64_t pts; 2274 status_t status = INVALID_OPERATION; 2275 2276 if (NULL != mOutput->stream->get_next_write_timestamp) { 2277 status = mOutput->stream->get_next_write_timestamp( 2278 mOutput->stream, &pts); 2279 } 2280 2281 if (status != NO_ERROR) { 2282 pts = AudioBufferProvider::kInvalidPTS; 2283 } 2284 2285 // mix buffers... 2286 mAudioMixer->process(pts); 2287 // increase sleep time progressively when application underrun condition clears. 2288 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2289 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2290 // such that we would underrun the audio HAL. 2291 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2292 sleepTimeShift--; 2293 } 2294 sleepTime = 0; 2295 standbyTime = systemTime() + standbyDelay; 2296 //TODO: delay standby when effects have a tail 2297} 2298 2299void AudioFlinger::MixerThread::threadLoop_sleepTime() 2300{ 2301 // If no tracks are ready, sleep once for the duration of an output 2302 // buffer size, then write 0s to the output 2303 if (sleepTime == 0) { 2304 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2305 sleepTime = activeSleepTime >> sleepTimeShift; 2306 if (sleepTime < kMinThreadSleepTimeUs) { 2307 sleepTime = kMinThreadSleepTimeUs; 2308 } 2309 // reduce sleep time in case of consecutive application underruns to avoid 2310 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2311 // duration we would end up writing less data than needed by the audio HAL if 2312 // the condition persists. 2313 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2314 sleepTimeShift++; 2315 } 2316 } else { 2317 sleepTime = idleSleepTime; 2318 } 2319 } else if (mBytesWritten != 0 || 2320 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2321 memset (mMixBuffer, 0, mixBufferSize); 2322 sleepTime = 0; 2323 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2324 } 2325 // TODO add standby time extension fct of effect tail 2326} 2327 2328// prepareTracks_l() must be called with ThreadBase::mLock held 2329AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2330 Vector< sp<Track> > *tracksToRemove) 2331{ 2332 2333 mixer_state mixerStatus = MIXER_IDLE; 2334 // find out which tracks need to be processed 2335 size_t count = mActiveTracks.size(); 2336 size_t mixedTracks = 0; 2337 size_t tracksWithEffect = 0; 2338 2339 float masterVolume = mMasterVolume; 2340 bool masterMute = mMasterMute; 2341 2342 if (masterMute) { 2343 masterVolume = 0; 2344 } 2345 // Delegate master volume control to effect in output mix effect chain if needed 2346 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2347 if (chain != 0) { 2348 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2349 chain->setVolume_l(&v, &v); 2350 masterVolume = (float)((v + (1 << 23)) >> 24); 2351 chain.clear(); 2352 } 2353 2354 for (size_t i=0 ; i<count ; i++) { 2355 sp<Track> t = mActiveTracks[i].promote(); 2356 if (t == 0) continue; 2357 2358 // this const just means the local variable doesn't change 2359 Track* const track = t.get(); 2360 audio_track_cblk_t* cblk = track->cblk(); 2361 2362 // The first time a track is added we wait 2363 // for all its buffers to be filled before processing it 2364 int name = track->name(); 2365 // make sure that we have enough frames to mix one full buffer. 2366 // enforce this condition only once to enable draining the buffer in case the client 2367 // app does not call stop() and relies on underrun to stop: 2368 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2369 // during last round 2370 uint32_t minFrames = 1; 2371 if (!track->isStopped() && !track->isPausing() && 2372 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2373 if (t->sampleRate() == (int)mSampleRate) { 2374 minFrames = mFrameCount; 2375 } else { 2376 // +1 for rounding and +1 for additional sample needed for interpolation 2377 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2378 // add frames already consumed but not yet released by the resampler 2379 // because cblk->framesReady() will include these frames 2380 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2381 // the minimum track buffer size is normally twice the number of frames necessary 2382 // to fill one buffer and the resampler should not leave more than one buffer worth 2383 // of unreleased frames after each pass, but just in case... 2384 ALOG_ASSERT(minFrames <= cblk->frameCount); 2385 } 2386 } 2387 if ((track->framesReady() >= minFrames) && track->isReady() && 2388 !track->isPaused() && !track->isTerminated()) 2389 { 2390 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2391 2392 mixedTracks++; 2393 2394 // track->mainBuffer() != mMixBuffer means there is an effect chain 2395 // connected to the track 2396 chain.clear(); 2397 if (track->mainBuffer() != mMixBuffer) { 2398 chain = getEffectChain_l(track->sessionId()); 2399 // Delegate volume control to effect in track effect chain if needed 2400 if (chain != 0) { 2401 tracksWithEffect++; 2402 } else { 2403 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2404 name, track->sessionId()); 2405 } 2406 } 2407 2408 2409 int param = AudioMixer::VOLUME; 2410 if (track->mFillingUpStatus == Track::FS_FILLED) { 2411 // no ramp for the first volume setting 2412 track->mFillingUpStatus = Track::FS_ACTIVE; 2413 if (track->mState == TrackBase::RESUMING) { 2414 track->mState = TrackBase::ACTIVE; 2415 param = AudioMixer::RAMP_VOLUME; 2416 } 2417 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2418 } else if (cblk->server != 0) { 2419 // If the track is stopped before the first frame was mixed, 2420 // do not apply ramp 2421 param = AudioMixer::RAMP_VOLUME; 2422 } 2423 2424 // compute volume for this track 2425 uint32_t vl, vr, va; 2426 if (track->isMuted() || track->isPausing() || 2427 mStreamTypes[track->streamType()].mute) { 2428 vl = vr = va = 0; 2429 if (track->isPausing()) { 2430 track->setPaused(); 2431 } 2432 } else { 2433 2434 // read original volumes with volume control 2435 float typeVolume = mStreamTypes[track->streamType()].volume; 2436 float v = masterVolume * typeVolume; 2437 uint32_t vlr = cblk->getVolumeLR(); 2438 vl = vlr & 0xFFFF; 2439 vr = vlr >> 16; 2440 // track volumes come from shared memory, so can't be trusted and must be clamped 2441 if (vl > MAX_GAIN_INT) { 2442 ALOGV("Track left volume out of range: %04X", vl); 2443 vl = MAX_GAIN_INT; 2444 } 2445 if (vr > MAX_GAIN_INT) { 2446 ALOGV("Track right volume out of range: %04X", vr); 2447 vr = MAX_GAIN_INT; 2448 } 2449 // now apply the master volume and stream type volume 2450 vl = (uint32_t)(v * vl) << 12; 2451 vr = (uint32_t)(v * vr) << 12; 2452 // assuming master volume and stream type volume each go up to 1.0, 2453 // vl and vr are now in 8.24 format 2454 2455 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2456 // send level comes from shared memory and so may be corrupt 2457 if (sendLevel > MAX_GAIN_INT) { 2458 ALOGV("Track send level out of range: %04X", sendLevel); 2459 sendLevel = MAX_GAIN_INT; 2460 } 2461 va = (uint32_t)(v * sendLevel); 2462 } 2463 // Delegate volume control to effect in track effect chain if needed 2464 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2465 // Do not ramp volume if volume is controlled by effect 2466 param = AudioMixer::VOLUME; 2467 track->mHasVolumeController = true; 2468 } else { 2469 // force no volume ramp when volume controller was just disabled or removed 2470 // from effect chain to avoid volume spike 2471 if (track->mHasVolumeController) { 2472 param = AudioMixer::VOLUME; 2473 } 2474 track->mHasVolumeController = false; 2475 } 2476 2477 // Convert volumes from 8.24 to 4.12 format 2478 // This additional clamping is needed in case chain->setVolume_l() overshot 2479 vl = (vl + (1 << 11)) >> 12; 2480 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2481 vr = (vr + (1 << 11)) >> 12; 2482 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2483 2484 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2485 2486 // XXX: these things DON'T need to be done each time 2487 mAudioMixer->setBufferProvider(name, track); 2488 mAudioMixer->enable(name); 2489 2490 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2491 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2492 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2493 mAudioMixer->setParameter( 2494 name, 2495 AudioMixer::TRACK, 2496 AudioMixer::FORMAT, (void *)track->format()); 2497 mAudioMixer->setParameter( 2498 name, 2499 AudioMixer::TRACK, 2500 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2501 mAudioMixer->setParameter( 2502 name, 2503 AudioMixer::RESAMPLE, 2504 AudioMixer::SAMPLE_RATE, 2505 (void *)(cblk->sampleRate)); 2506 mAudioMixer->setParameter( 2507 name, 2508 AudioMixer::TRACK, 2509 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2510 mAudioMixer->setParameter( 2511 name, 2512 AudioMixer::TRACK, 2513 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2514 2515 // reset retry count 2516 track->mRetryCount = kMaxTrackRetries; 2517 2518 // If one track is ready, set the mixer ready if: 2519 // - the mixer was not ready during previous round OR 2520 // - no other track is not ready 2521 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2522 mixerStatus != MIXER_TRACKS_ENABLED) { 2523 mixerStatus = MIXER_TRACKS_READY; 2524 } 2525 } else { 2526 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2527 if (track->isStopped()) { 2528 track->reset(); 2529 } 2530 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2531 // We have consumed all the buffers of this track. 2532 // Remove it from the list of active tracks. 2533 tracksToRemove->add(track); 2534 } else { 2535 // No buffers for this track. Give it a few chances to 2536 // fill a buffer, then remove it from active list. 2537 if (--(track->mRetryCount) <= 0) { 2538 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2539 tracksToRemove->add(track); 2540 // indicate to client process that the track was disabled because of underrun 2541 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2542 // If one track is not ready, mark the mixer also not ready if: 2543 // - the mixer was ready during previous round OR 2544 // - no other track is ready 2545 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2546 mixerStatus != MIXER_TRACKS_READY) { 2547 mixerStatus = MIXER_TRACKS_ENABLED; 2548 } 2549 } 2550 mAudioMixer->disable(name); 2551 } 2552 } 2553 2554 // remove all the tracks that need to be... 2555 count = tracksToRemove->size(); 2556 if (CC_UNLIKELY(count)) { 2557 for (size_t i=0 ; i<count ; i++) { 2558 const sp<Track>& track = tracksToRemove->itemAt(i); 2559 mActiveTracks.remove(track); 2560 if (track->mainBuffer() != mMixBuffer) { 2561 chain = getEffectChain_l(track->sessionId()); 2562 if (chain != 0) { 2563 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2564 chain->decActiveTrackCnt(); 2565 } 2566 } 2567 if (track->isTerminated()) { 2568 removeTrack_l(track); 2569 } 2570 } 2571 } 2572 2573 // mix buffer must be cleared if all tracks are connected to an 2574 // effect chain as in this case the mixer will not write to 2575 // mix buffer and track effects will accumulate into it 2576 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2577 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2578 } 2579 2580 return mixerStatus; 2581} 2582 2583/* 2584The derived values that are cached: 2585 - mixBufferSize from frame count * frame size 2586 - activeSleepTime from activeSleepTimeUs() 2587 - idleSleepTime from idleSleepTimeUs() 2588 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2589 - maxPeriod from frame count and sample rate (MIXER only) 2590 2591The parameters that affect these derived values are: 2592 - frame count 2593 - frame size 2594 - sample rate 2595 - device type: A2DP or not 2596 - device latency 2597 - format: PCM or not 2598 - active sleep time 2599 - idle sleep time 2600*/ 2601 2602void AudioFlinger::PlaybackThread::cacheParameters_l() 2603{ 2604 mixBufferSize = mFrameCount * mFrameSize; 2605 activeSleepTime = activeSleepTimeUs(); 2606 idleSleepTime = idleSleepTimeUs(); 2607} 2608 2609void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2610{ 2611 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2612 this, streamType, mTracks.size()); 2613 Mutex::Autolock _l(mLock); 2614 2615 size_t size = mTracks.size(); 2616 for (size_t i = 0; i < size; i++) { 2617 sp<Track> t = mTracks[i]; 2618 if (t->streamType() == streamType) { 2619 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2620 t->mCblk->cv.signal(); 2621 } 2622 } 2623} 2624 2625void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2626{ 2627 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2628 this, streamType, valid); 2629 Mutex::Autolock _l(mLock); 2630 2631 mStreamTypes[streamType].valid = valid; 2632} 2633 2634// getTrackName_l() must be called with ThreadBase::mLock held 2635int AudioFlinger::MixerThread::getTrackName_l() 2636{ 2637 return mAudioMixer->getTrackName(); 2638} 2639 2640// deleteTrackName_l() must be called with ThreadBase::mLock held 2641void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2642{ 2643 ALOGV("remove track (%d) and delete from mixer", name); 2644 mAudioMixer->deleteTrackName(name); 2645} 2646 2647// checkForNewParameters_l() must be called with ThreadBase::mLock held 2648bool AudioFlinger::MixerThread::checkForNewParameters_l() 2649{ 2650 bool reconfig = false; 2651 2652 while (!mNewParameters.isEmpty()) { 2653 status_t status = NO_ERROR; 2654 String8 keyValuePair = mNewParameters[0]; 2655 AudioParameter param = AudioParameter(keyValuePair); 2656 int value; 2657 2658 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2659 reconfig = true; 2660 } 2661 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2662 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2663 status = BAD_VALUE; 2664 } else { 2665 reconfig = true; 2666 } 2667 } 2668 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2669 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2670 status = BAD_VALUE; 2671 } else { 2672 reconfig = true; 2673 } 2674 } 2675 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2676 // do not accept frame count changes if tracks are open as the track buffer 2677 // size depends on frame count and correct behavior would not be guaranteed 2678 // if frame count is changed after track creation 2679 if (!mTracks.isEmpty()) { 2680 status = INVALID_OPERATION; 2681 } else { 2682 reconfig = true; 2683 } 2684 } 2685 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2686#ifdef ADD_BATTERY_DATA 2687 // when changing the audio output device, call addBatteryData to notify 2688 // the change 2689 if ((int)mDevice != value) { 2690 uint32_t params = 0; 2691 // check whether speaker is on 2692 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2693 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2694 } 2695 2696 int deviceWithoutSpeaker 2697 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2698 // check if any other device (except speaker) is on 2699 if (value & deviceWithoutSpeaker ) { 2700 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2701 } 2702 2703 if (params != 0) { 2704 addBatteryData(params); 2705 } 2706 } 2707#endif 2708 2709 // forward device change to effects that have requested to be 2710 // aware of attached audio device. 2711 mDevice = (uint32_t)value; 2712 for (size_t i = 0; i < mEffectChains.size(); i++) { 2713 mEffectChains[i]->setDevice_l(mDevice); 2714 } 2715 } 2716 2717 if (status == NO_ERROR) { 2718 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2719 keyValuePair.string()); 2720 if (!mStandby && status == INVALID_OPERATION) { 2721 mOutput->stream->common.standby(&mOutput->stream->common); 2722 mStandby = true; 2723 mBytesWritten = 0; 2724 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2725 keyValuePair.string()); 2726 } 2727 if (status == NO_ERROR && reconfig) { 2728 delete mAudioMixer; 2729 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2730 mAudioMixer = NULL; 2731 readOutputParameters(); 2732 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2733 for (size_t i = 0; i < mTracks.size() ; i++) { 2734 int name = getTrackName_l(); 2735 if (name < 0) break; 2736 mTracks[i]->mName = name; 2737 // limit track sample rate to 2 x new output sample rate 2738 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2739 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2740 } 2741 } 2742 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2743 } 2744 } 2745 2746 mNewParameters.removeAt(0); 2747 2748 mParamStatus = status; 2749 mParamCond.signal(); 2750 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2751 // already timed out waiting for the status and will never signal the condition. 2752 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2753 } 2754 return reconfig; 2755} 2756 2757status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2758{ 2759 const size_t SIZE = 256; 2760 char buffer[SIZE]; 2761 String8 result; 2762 2763 PlaybackThread::dumpInternals(fd, args); 2764 2765 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2766 result.append(buffer); 2767 write(fd, result.string(), result.size()); 2768 return NO_ERROR; 2769} 2770 2771uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2772{ 2773 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2774} 2775 2776uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2777{ 2778 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2779} 2780 2781void AudioFlinger::MixerThread::cacheParameters_l() 2782{ 2783 PlaybackThread::cacheParameters_l(); 2784 2785 // FIXME: Relaxed timing because of a certain device that can't meet latency 2786 // Should be reduced to 2x after the vendor fixes the driver issue 2787 // increase threshold again due to low power audio mode. The way this warning 2788 // threshold is calculated and its usefulness should be reconsidered anyway. 2789 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2790} 2791 2792// ---------------------------------------------------------------------------- 2793AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2794 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2795 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2796 // mLeftVolFloat, mRightVolFloat 2797 // mLeftVolShort, mRightVolShort 2798{ 2799} 2800 2801AudioFlinger::DirectOutputThread::~DirectOutputThread() 2802{ 2803} 2804 2805AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2806 Vector< sp<Track> > *tracksToRemove 2807) 2808{ 2809 sp<Track> trackToRemove; 2810 2811 mixer_state mixerStatus = MIXER_IDLE; 2812 2813 // find out which tracks need to be processed 2814 if (mActiveTracks.size() != 0) { 2815 sp<Track> t = mActiveTracks[0].promote(); 2816 // The track died recently 2817 if (t == 0) return MIXER_IDLE; 2818 2819 Track* const track = t.get(); 2820 audio_track_cblk_t* cblk = track->cblk(); 2821 2822 // The first time a track is added we wait 2823 // for all its buffers to be filled before processing it 2824 if (cblk->framesReady() && track->isReady() && 2825 !track->isPaused() && !track->isTerminated()) 2826 { 2827 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2828 2829 if (track->mFillingUpStatus == Track::FS_FILLED) { 2830 track->mFillingUpStatus = Track::FS_ACTIVE; 2831 mLeftVolFloat = mRightVolFloat = 0; 2832 mLeftVolShort = mRightVolShort = 0; 2833 if (track->mState == TrackBase::RESUMING) { 2834 track->mState = TrackBase::ACTIVE; 2835 rampVolume = true; 2836 } 2837 } else if (cblk->server != 0) { 2838 // If the track is stopped before the first frame was mixed, 2839 // do not apply ramp 2840 rampVolume = true; 2841 } 2842 // compute volume for this track 2843 float left, right; 2844 if (track->isMuted() || mMasterMute || track->isPausing() || 2845 mStreamTypes[track->streamType()].mute) { 2846 left = right = 0; 2847 if (track->isPausing()) { 2848 track->setPaused(); 2849 } 2850 } else { 2851 float typeVolume = mStreamTypes[track->streamType()].volume; 2852 float v = mMasterVolume * typeVolume; 2853 uint32_t vlr = cblk->getVolumeLR(); 2854 float v_clamped = v * (vlr & 0xFFFF); 2855 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2856 left = v_clamped/MAX_GAIN; 2857 v_clamped = v * (vlr >> 16); 2858 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2859 right = v_clamped/MAX_GAIN; 2860 } 2861 2862 if (left != mLeftVolFloat || right != mRightVolFloat) { 2863 mLeftVolFloat = left; 2864 mRightVolFloat = right; 2865 2866 // If audio HAL implements volume control, 2867 // force software volume to nominal value 2868 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2869 left = 1.0f; 2870 right = 1.0f; 2871 } 2872 2873 // Convert volumes from float to 8.24 2874 uint32_t vl = (uint32_t)(left * (1 << 24)); 2875 uint32_t vr = (uint32_t)(right * (1 << 24)); 2876 2877 // Delegate volume control to effect in track effect chain if needed 2878 // only one effect chain can be present on DirectOutputThread, so if 2879 // there is one, the track is connected to it 2880 if (!mEffectChains.isEmpty()) { 2881 // Do not ramp volume if volume is controlled by effect 2882 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2883 rampVolume = false; 2884 } 2885 } 2886 2887 // Convert volumes from 8.24 to 4.12 format 2888 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2889 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2890 leftVol = (uint16_t)v_clamped; 2891 v_clamped = (vr + (1 << 11)) >> 12; 2892 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2893 rightVol = (uint16_t)v_clamped; 2894 } else { 2895 leftVol = mLeftVolShort; 2896 rightVol = mRightVolShort; 2897 rampVolume = false; 2898 } 2899 2900 // reset retry count 2901 track->mRetryCount = kMaxTrackRetriesDirect; 2902 mActiveTrack = t; 2903 mixerStatus = MIXER_TRACKS_READY; 2904 } else { 2905 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2906 if (track->isStopped()) { 2907 track->reset(); 2908 } 2909 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2910 // We have consumed all the buffers of this track. 2911 // Remove it from the list of active tracks. 2912 trackToRemove = track; 2913 } else { 2914 // No buffers for this track. Give it a few chances to 2915 // fill a buffer, then remove it from active list. 2916 if (--(track->mRetryCount) <= 0) { 2917 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2918 trackToRemove = track; 2919 } else { 2920 mixerStatus = MIXER_TRACKS_ENABLED; 2921 } 2922 } 2923 } 2924 } 2925 2926 // FIXME merge this with similar code for removing multiple tracks 2927 // remove all the tracks that need to be... 2928 if (CC_UNLIKELY(trackToRemove != 0)) { 2929 tracksToRemove->add(trackToRemove); 2930 mActiveTracks.remove(trackToRemove); 2931 if (!mEffectChains.isEmpty()) { 2932 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2933 trackToRemove->sessionId()); 2934 mEffectChains[0]->decActiveTrackCnt(); 2935 } 2936 if (trackToRemove->isTerminated()) { 2937 removeTrack_l(trackToRemove); 2938 } 2939 } 2940 2941 return mixerStatus; 2942} 2943 2944void AudioFlinger::DirectOutputThread::threadLoop_mix() 2945{ 2946 AudioBufferProvider::Buffer buffer; 2947 size_t frameCount = mFrameCount; 2948 int8_t *curBuf = (int8_t *)mMixBuffer; 2949 // output audio to hardware 2950 while (frameCount) { 2951 buffer.frameCount = frameCount; 2952 mActiveTrack->getNextBuffer(&buffer); 2953 if (CC_UNLIKELY(buffer.raw == NULL)) { 2954 memset(curBuf, 0, frameCount * mFrameSize); 2955 break; 2956 } 2957 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2958 frameCount -= buffer.frameCount; 2959 curBuf += buffer.frameCount * mFrameSize; 2960 mActiveTrack->releaseBuffer(&buffer); 2961 } 2962 sleepTime = 0; 2963 standbyTime = systemTime() + standbyDelay; 2964 mActiveTrack.clear(); 2965 2966 // apply volume 2967 2968 // Do not apply volume on compressed audio 2969 if (!audio_is_linear_pcm(mFormat)) { 2970 return; 2971 } 2972 2973 // convert to signed 16 bit before volume calculation 2974 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2975 size_t count = mFrameCount * mChannelCount; 2976 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2977 int16_t *dst = mMixBuffer + count-1; 2978 while (count--) { 2979 *dst-- = (int16_t)(*src--^0x80) << 8; 2980 } 2981 } 2982 2983 frameCount = mFrameCount; 2984 int16_t *out = mMixBuffer; 2985 if (rampVolume) { 2986 if (mChannelCount == 1) { 2987 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2988 int32_t vlInc = d / (int32_t)frameCount; 2989 int32_t vl = ((int32_t)mLeftVolShort << 16); 2990 do { 2991 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2992 out++; 2993 vl += vlInc; 2994 } while (--frameCount); 2995 2996 } else { 2997 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2998 int32_t vlInc = d / (int32_t)frameCount; 2999 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3000 int32_t vrInc = d / (int32_t)frameCount; 3001 int32_t vl = ((int32_t)mLeftVolShort << 16); 3002 int32_t vr = ((int32_t)mRightVolShort << 16); 3003 do { 3004 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3005 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3006 out += 2; 3007 vl += vlInc; 3008 vr += vrInc; 3009 } while (--frameCount); 3010 } 3011 } else { 3012 if (mChannelCount == 1) { 3013 do { 3014 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3015 out++; 3016 } while (--frameCount); 3017 } else { 3018 do { 3019 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3020 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3021 out += 2; 3022 } while (--frameCount); 3023 } 3024 } 3025 3026 // convert back to unsigned 8 bit after volume calculation 3027 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3028 size_t count = mFrameCount * mChannelCount; 3029 int16_t *src = mMixBuffer; 3030 uint8_t *dst = (uint8_t *)mMixBuffer; 3031 while (count--) { 3032 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3033 } 3034 } 3035 3036 mLeftVolShort = leftVol; 3037 mRightVolShort = rightVol; 3038} 3039 3040void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3041{ 3042 if (sleepTime == 0) { 3043 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3044 sleepTime = activeSleepTime; 3045 } else { 3046 sleepTime = idleSleepTime; 3047 } 3048 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3049 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3050 sleepTime = 0; 3051 } 3052} 3053 3054// getTrackName_l() must be called with ThreadBase::mLock held 3055int AudioFlinger::DirectOutputThread::getTrackName_l() 3056{ 3057 return 0; 3058} 3059 3060// deleteTrackName_l() must be called with ThreadBase::mLock held 3061void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3062{ 3063} 3064 3065// checkForNewParameters_l() must be called with ThreadBase::mLock held 3066bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3067{ 3068 bool reconfig = false; 3069 3070 while (!mNewParameters.isEmpty()) { 3071 status_t status = NO_ERROR; 3072 String8 keyValuePair = mNewParameters[0]; 3073 AudioParameter param = AudioParameter(keyValuePair); 3074 int value; 3075 3076 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3077 // do not accept frame count changes if tracks are open as the track buffer 3078 // size depends on frame count and correct behavior would not be garantied 3079 // if frame count is changed after track creation 3080 if (!mTracks.isEmpty()) { 3081 status = INVALID_OPERATION; 3082 } else { 3083 reconfig = true; 3084 } 3085 } 3086 if (status == NO_ERROR) { 3087 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3088 keyValuePair.string()); 3089 if (!mStandby && status == INVALID_OPERATION) { 3090 mOutput->stream->common.standby(&mOutput->stream->common); 3091 mStandby = true; 3092 mBytesWritten = 0; 3093 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3094 keyValuePair.string()); 3095 } 3096 if (status == NO_ERROR && reconfig) { 3097 readOutputParameters(); 3098 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3099 } 3100 } 3101 3102 mNewParameters.removeAt(0); 3103 3104 mParamStatus = status; 3105 mParamCond.signal(); 3106 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3107 // already timed out waiting for the status and will never signal the condition. 3108 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3109 } 3110 return reconfig; 3111} 3112 3113uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3114{ 3115 uint32_t time; 3116 if (audio_is_linear_pcm(mFormat)) { 3117 time = PlaybackThread::activeSleepTimeUs(); 3118 } else { 3119 time = 10000; 3120 } 3121 return time; 3122} 3123 3124uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3125{ 3126 uint32_t time; 3127 if (audio_is_linear_pcm(mFormat)) { 3128 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3129 } else { 3130 time = 10000; 3131 } 3132 return time; 3133} 3134 3135uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3136{ 3137 uint32_t time; 3138 if (audio_is_linear_pcm(mFormat)) { 3139 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3140 } else { 3141 time = 10000; 3142 } 3143 return time; 3144} 3145 3146void AudioFlinger::DirectOutputThread::cacheParameters_l() 3147{ 3148 PlaybackThread::cacheParameters_l(); 3149 3150 // use shorter standby delay as on normal output to release 3151 // hardware resources as soon as possible 3152 standbyDelay = microseconds(activeSleepTime*2); 3153} 3154 3155// ---------------------------------------------------------------------------- 3156 3157AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3158 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3159 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3160 mWaitTimeMs(UINT_MAX) 3161{ 3162 addOutputTrack(mainThread); 3163} 3164 3165AudioFlinger::DuplicatingThread::~DuplicatingThread() 3166{ 3167 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3168 mOutputTracks[i]->destroy(); 3169 } 3170} 3171 3172void AudioFlinger::DuplicatingThread::threadLoop_mix() 3173{ 3174 // mix buffers... 3175 if (outputsReady(outputTracks)) { 3176 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3177 } else { 3178 memset(mMixBuffer, 0, mixBufferSize); 3179 } 3180 sleepTime = 0; 3181 writeFrames = mFrameCount; 3182} 3183 3184void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3185{ 3186 if (sleepTime == 0) { 3187 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3188 sleepTime = activeSleepTime; 3189 } else { 3190 sleepTime = idleSleepTime; 3191 } 3192 } else if (mBytesWritten != 0) { 3193 // flush remaining overflow buffers in output tracks 3194 for (size_t i = 0; i < outputTracks.size(); i++) { 3195 if (outputTracks[i]->isActive()) { 3196 sleepTime = 0; 3197 writeFrames = 0; 3198 memset(mMixBuffer, 0, mixBufferSize); 3199 break; 3200 } 3201 } 3202 } 3203} 3204 3205void AudioFlinger::DuplicatingThread::threadLoop_write() 3206{ 3207 standbyTime = systemTime() + standbyDelay; 3208 for (size_t i = 0; i < outputTracks.size(); i++) { 3209 outputTracks[i]->write(mMixBuffer, writeFrames); 3210 } 3211 mBytesWritten += mixBufferSize; 3212} 3213 3214void AudioFlinger::DuplicatingThread::threadLoop_standby() 3215{ 3216 // DuplicatingThread implements standby by stopping all tracks 3217 for (size_t i = 0; i < outputTracks.size(); i++) { 3218 outputTracks[i]->stop(); 3219 } 3220} 3221 3222void AudioFlinger::DuplicatingThread::saveOutputTracks() 3223{ 3224 outputTracks = mOutputTracks; 3225} 3226 3227void AudioFlinger::DuplicatingThread::clearOutputTracks() 3228{ 3229 outputTracks.clear(); 3230} 3231 3232void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3233{ 3234 Mutex::Autolock _l(mLock); 3235 // FIXME explain this formula 3236 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3237 OutputTrack *outputTrack = new OutputTrack(thread, 3238 this, 3239 mSampleRate, 3240 mFormat, 3241 mChannelMask, 3242 frameCount); 3243 if (outputTrack->cblk() != NULL) { 3244 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3245 mOutputTracks.add(outputTrack); 3246 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3247 updateWaitTime_l(); 3248 } 3249} 3250 3251void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3252{ 3253 Mutex::Autolock _l(mLock); 3254 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3255 if (mOutputTracks[i]->thread() == thread) { 3256 mOutputTracks[i]->destroy(); 3257 mOutputTracks.removeAt(i); 3258 updateWaitTime_l(); 3259 return; 3260 } 3261 } 3262 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3263} 3264 3265// caller must hold mLock 3266void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3267{ 3268 mWaitTimeMs = UINT_MAX; 3269 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3270 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3271 if (strong != 0) { 3272 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3273 if (waitTimeMs < mWaitTimeMs) { 3274 mWaitTimeMs = waitTimeMs; 3275 } 3276 } 3277 } 3278} 3279 3280 3281bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3282{ 3283 for (size_t i = 0; i < outputTracks.size(); i++) { 3284 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3285 if (thread == 0) { 3286 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3287 return false; 3288 } 3289 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3290 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3291 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3292 return false; 3293 } 3294 } 3295 return true; 3296} 3297 3298uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3299{ 3300 return (mWaitTimeMs * 1000) / 2; 3301} 3302 3303void AudioFlinger::DuplicatingThread::cacheParameters_l() 3304{ 3305 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3306 updateWaitTime_l(); 3307 3308 MixerThread::cacheParameters_l(); 3309} 3310 3311// ---------------------------------------------------------------------------- 3312 3313// TrackBase constructor must be called with AudioFlinger::mLock held 3314AudioFlinger::ThreadBase::TrackBase::TrackBase( 3315 ThreadBase *thread, 3316 const sp<Client>& client, 3317 uint32_t sampleRate, 3318 audio_format_t format, 3319 uint32_t channelMask, 3320 int frameCount, 3321 const sp<IMemory>& sharedBuffer, 3322 int sessionId) 3323 : RefBase(), 3324 mThread(thread), 3325 mClient(client), 3326 mCblk(NULL), 3327 // mBuffer 3328 // mBufferEnd 3329 mFrameCount(0), 3330 mState(IDLE), 3331 mFormat(format), 3332 mStepServerFailed(false), 3333 mSessionId(sessionId) 3334 // mChannelCount 3335 // mChannelMask 3336{ 3337 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3338 3339 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3340 size_t size = sizeof(audio_track_cblk_t); 3341 uint8_t channelCount = popcount(channelMask); 3342 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3343 if (sharedBuffer == 0) { 3344 size += bufferSize; 3345 } 3346 3347 if (client != NULL) { 3348 mCblkMemory = client->heap()->allocate(size); 3349 if (mCblkMemory != 0) { 3350 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3351 if (mCblk != NULL) { // construct the shared structure in-place. 3352 new(mCblk) audio_track_cblk_t(); 3353 // clear all buffers 3354 mCblk->frameCount = frameCount; 3355 mCblk->sampleRate = sampleRate; 3356 mChannelCount = channelCount; 3357 mChannelMask = channelMask; 3358 if (sharedBuffer == 0) { 3359 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3360 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3361 // Force underrun condition to avoid false underrun callback until first data is 3362 // written to buffer (other flags are cleared) 3363 mCblk->flags = CBLK_UNDERRUN_ON; 3364 } else { 3365 mBuffer = sharedBuffer->pointer(); 3366 } 3367 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3368 } 3369 } else { 3370 ALOGE("not enough memory for AudioTrack size=%u", size); 3371 client->heap()->dump("AudioTrack"); 3372 return; 3373 } 3374 } else { 3375 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3376 // construct the shared structure in-place. 3377 new(mCblk) audio_track_cblk_t(); 3378 // clear all buffers 3379 mCblk->frameCount = frameCount; 3380 mCblk->sampleRate = sampleRate; 3381 mChannelCount = channelCount; 3382 mChannelMask = channelMask; 3383 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3384 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3385 // Force underrun condition to avoid false underrun callback until first data is 3386 // written to buffer (other flags are cleared) 3387 mCblk->flags = CBLK_UNDERRUN_ON; 3388 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3389 } 3390} 3391 3392AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3393{ 3394 if (mCblk != NULL) { 3395 if (mClient == 0) { 3396 delete mCblk; 3397 } else { 3398 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3399 } 3400 } 3401 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3402 if (mClient != 0) { 3403 // Client destructor must run with AudioFlinger mutex locked 3404 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3405 // If the client's reference count drops to zero, the associated destructor 3406 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3407 // relying on the automatic clear() at end of scope. 3408 mClient.clear(); 3409 } 3410} 3411 3412// AudioBufferProvider interface 3413// getNextBuffer() = 0; 3414// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3415void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3416{ 3417 buffer->raw = NULL; 3418 mFrameCount = buffer->frameCount; 3419 (void) step(); // ignore return value of step() 3420 buffer->frameCount = 0; 3421} 3422 3423bool AudioFlinger::ThreadBase::TrackBase::step() { 3424 bool result; 3425 audio_track_cblk_t* cblk = this->cblk(); 3426 3427 result = cblk->stepServer(mFrameCount); 3428 if (!result) { 3429 ALOGV("stepServer failed acquiring cblk mutex"); 3430 mStepServerFailed = true; 3431 } 3432 return result; 3433} 3434 3435void AudioFlinger::ThreadBase::TrackBase::reset() { 3436 audio_track_cblk_t* cblk = this->cblk(); 3437 3438 cblk->user = 0; 3439 cblk->server = 0; 3440 cblk->userBase = 0; 3441 cblk->serverBase = 0; 3442 mStepServerFailed = false; 3443 ALOGV("TrackBase::reset"); 3444} 3445 3446int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3447 return (int)mCblk->sampleRate; 3448} 3449 3450void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3451 audio_track_cblk_t* cblk = this->cblk(); 3452 size_t frameSize = cblk->frameSize; 3453 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3454 int8_t *bufferEnd = bufferStart + frames * frameSize; 3455 3456 // Check validity of returned pointer in case the track control block would have been corrupted. 3457 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3458 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3459 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3460 server %d, serverBase %d, user %d, userBase %d", 3461 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3462 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3463 return NULL; 3464 } 3465 3466 return bufferStart; 3467} 3468 3469// ---------------------------------------------------------------------------- 3470 3471// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3472AudioFlinger::PlaybackThread::Track::Track( 3473 PlaybackThread *thread, 3474 const sp<Client>& client, 3475 audio_stream_type_t streamType, 3476 uint32_t sampleRate, 3477 audio_format_t format, 3478 uint32_t channelMask, 3479 int frameCount, 3480 const sp<IMemory>& sharedBuffer, 3481 int sessionId) 3482 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3483 mMute(false), 3484 // mFillingUpStatus ? 3485 // mRetryCount initialized later when needed 3486 mSharedBuffer(sharedBuffer), 3487 mStreamType(streamType), 3488 mName(-1), // see note below 3489 mMainBuffer(thread->mixBuffer()), 3490 mAuxBuffer(NULL), 3491 mAuxEffectId(0), mHasVolumeController(false) 3492{ 3493 if (mCblk != NULL) { 3494 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3495 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3496 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3497 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3498 mName = thread->getTrackName_l(); 3499 if (mName < 0) { 3500 ALOGE("no more track names available"); 3501 } 3502 } 3503 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3504} 3505 3506AudioFlinger::PlaybackThread::Track::~Track() 3507{ 3508 ALOGV("PlaybackThread::Track destructor"); 3509 sp<ThreadBase> thread = mThread.promote(); 3510 if (thread != 0) { 3511 Mutex::Autolock _l(thread->mLock); 3512 mState = TERMINATED; 3513 } 3514} 3515 3516void AudioFlinger::PlaybackThread::Track::destroy() 3517{ 3518 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3519 // by removing it from mTracks vector, so there is a risk that this Tracks's 3520 // destructor is called. As the destructor needs to lock mLock, 3521 // we must acquire a strong reference on this Track before locking mLock 3522 // here so that the destructor is called only when exiting this function. 3523 // On the other hand, as long as Track::destroy() is only called by 3524 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3525 // this Track with its member mTrack. 3526 sp<Track> keep(this); 3527 { // scope for mLock 3528 sp<ThreadBase> thread = mThread.promote(); 3529 if (thread != 0) { 3530 if (!isOutputTrack()) { 3531 if (mState == ACTIVE || mState == RESUMING) { 3532 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3533 3534#ifdef ADD_BATTERY_DATA 3535 // to track the speaker usage 3536 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3537#endif 3538 } 3539 AudioSystem::releaseOutput(thread->id()); 3540 } 3541 Mutex::Autolock _l(thread->mLock); 3542 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3543 playbackThread->destroyTrack_l(this); 3544 } 3545 } 3546} 3547 3548void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3549{ 3550 uint32_t vlr = mCblk->getVolumeLR(); 3551 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", 3552 mName - AudioMixer::TRACK0, 3553 (mClient == 0) ? getpid_cached : mClient->pid(), 3554 mStreamType, 3555 mFormat, 3556 mChannelMask, 3557 mSessionId, 3558 mFrameCount, 3559 mState, 3560 mMute, 3561 mFillingUpStatus, 3562 mCblk->sampleRate, 3563 vlr & 0xFFFF, 3564 vlr >> 16, 3565 mCblk->server, 3566 mCblk->user, 3567 (int)mMainBuffer, 3568 (int)mAuxBuffer); 3569} 3570 3571// AudioBufferProvider interface 3572status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3573 AudioBufferProvider::Buffer* buffer, int64_t pts) 3574{ 3575 audio_track_cblk_t* cblk = this->cblk(); 3576 uint32_t framesReady; 3577 uint32_t framesReq = buffer->frameCount; 3578 3579 // Check if last stepServer failed, try to step now 3580 if (mStepServerFailed) { 3581 if (!step()) goto getNextBuffer_exit; 3582 ALOGV("stepServer recovered"); 3583 mStepServerFailed = false; 3584 } 3585 3586 framesReady = cblk->framesReady(); 3587 3588 if (CC_LIKELY(framesReady)) { 3589 uint32_t s = cblk->server; 3590 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3591 3592 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3593 if (framesReq > framesReady) { 3594 framesReq = framesReady; 3595 } 3596 if (s + framesReq > bufferEnd) { 3597 framesReq = bufferEnd - s; 3598 } 3599 3600 buffer->raw = getBuffer(s, framesReq); 3601 if (buffer->raw == NULL) goto getNextBuffer_exit; 3602 3603 buffer->frameCount = framesReq; 3604 return NO_ERROR; 3605 } 3606 3607getNextBuffer_exit: 3608 buffer->raw = NULL; 3609 buffer->frameCount = 0; 3610 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3611 return NOT_ENOUGH_DATA; 3612} 3613 3614uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3615 return mCblk->framesReady(); 3616} 3617 3618bool AudioFlinger::PlaybackThread::Track::isReady() const { 3619 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3620 3621 if (framesReady() >= mCblk->frameCount || 3622 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3623 mFillingUpStatus = FS_FILLED; 3624 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3625 return true; 3626 } 3627 return false; 3628} 3629 3630status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3631{ 3632 status_t status = NO_ERROR; 3633 ALOGV("start(%d), calling pid %d session %d tid %d", 3634 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3635 sp<ThreadBase> thread = mThread.promote(); 3636 if (thread != 0) { 3637 Mutex::Autolock _l(thread->mLock); 3638 track_state state = mState; 3639 // here the track could be either new, or restarted 3640 // in both cases "unstop" the track 3641 if (mState == PAUSED) { 3642 mState = TrackBase::RESUMING; 3643 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3644 } else { 3645 mState = TrackBase::ACTIVE; 3646 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3647 } 3648 3649 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3650 thread->mLock.unlock(); 3651 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3652 thread->mLock.lock(); 3653 3654#ifdef ADD_BATTERY_DATA 3655 // to track the speaker usage 3656 if (status == NO_ERROR) { 3657 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3658 } 3659#endif 3660 } 3661 if (status == NO_ERROR) { 3662 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3663 playbackThread->addTrack_l(this); 3664 } else { 3665 mState = state; 3666 } 3667 } else { 3668 status = BAD_VALUE; 3669 } 3670 return status; 3671} 3672 3673void AudioFlinger::PlaybackThread::Track::stop() 3674{ 3675 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3676 sp<ThreadBase> thread = mThread.promote(); 3677 if (thread != 0) { 3678 Mutex::Autolock _l(thread->mLock); 3679 track_state state = mState; 3680 if (mState > STOPPED) { 3681 mState = STOPPED; 3682 // If the track is not active (PAUSED and buffers full), flush buffers 3683 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3684 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3685 reset(); 3686 } 3687 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3688 } 3689 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3690 thread->mLock.unlock(); 3691 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3692 thread->mLock.lock(); 3693 3694#ifdef ADD_BATTERY_DATA 3695 // to track the speaker usage 3696 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3697#endif 3698 } 3699 } 3700} 3701 3702void AudioFlinger::PlaybackThread::Track::pause() 3703{ 3704 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3705 sp<ThreadBase> thread = mThread.promote(); 3706 if (thread != 0) { 3707 Mutex::Autolock _l(thread->mLock); 3708 if (mState == ACTIVE || mState == RESUMING) { 3709 mState = PAUSING; 3710 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3711 if (!isOutputTrack()) { 3712 thread->mLock.unlock(); 3713 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3714 thread->mLock.lock(); 3715 3716#ifdef ADD_BATTERY_DATA 3717 // to track the speaker usage 3718 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3719#endif 3720 } 3721 } 3722 } 3723} 3724 3725void AudioFlinger::PlaybackThread::Track::flush() 3726{ 3727 ALOGV("flush(%d)", mName); 3728 sp<ThreadBase> thread = mThread.promote(); 3729 if (thread != 0) { 3730 Mutex::Autolock _l(thread->mLock); 3731 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3732 return; 3733 } 3734 // No point remaining in PAUSED state after a flush => go to 3735 // STOPPED state 3736 mState = STOPPED; 3737 3738 // do not reset the track if it is still in the process of being stopped or paused. 3739 // this will be done by prepareTracks_l() when the track is stopped. 3740 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3741 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3742 reset(); 3743 } 3744 } 3745} 3746 3747void AudioFlinger::PlaybackThread::Track::reset() 3748{ 3749 // Do not reset twice to avoid discarding data written just after a flush and before 3750 // the audioflinger thread detects the track is stopped. 3751 if (!mResetDone) { 3752 TrackBase::reset(); 3753 // Force underrun condition to avoid false underrun callback until first data is 3754 // written to buffer 3755 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3756 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3757 mFillingUpStatus = FS_FILLING; 3758 mResetDone = true; 3759 } 3760} 3761 3762void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3763{ 3764 mMute = muted; 3765} 3766 3767status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3768{ 3769 status_t status = DEAD_OBJECT; 3770 sp<ThreadBase> thread = mThread.promote(); 3771 if (thread != 0) { 3772 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3773 status = playbackThread->attachAuxEffect(this, EffectId); 3774 } 3775 return status; 3776} 3777 3778void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3779{ 3780 mAuxEffectId = EffectId; 3781 mAuxBuffer = buffer; 3782} 3783 3784// timed audio tracks 3785 3786sp<AudioFlinger::PlaybackThread::TimedTrack> 3787AudioFlinger::PlaybackThread::TimedTrack::create( 3788 PlaybackThread *thread, 3789 const sp<Client>& client, 3790 audio_stream_type_t streamType, 3791 uint32_t sampleRate, 3792 audio_format_t format, 3793 uint32_t channelMask, 3794 int frameCount, 3795 const sp<IMemory>& sharedBuffer, 3796 int sessionId) { 3797 if (!client->reserveTimedTrack()) 3798 return NULL; 3799 3800 return new TimedTrack( 3801 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3802 sharedBuffer, sessionId); 3803} 3804 3805AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3806 PlaybackThread *thread, 3807 const sp<Client>& client, 3808 audio_stream_type_t streamType, 3809 uint32_t sampleRate, 3810 audio_format_t format, 3811 uint32_t channelMask, 3812 int frameCount, 3813 const sp<IMemory>& sharedBuffer, 3814 int sessionId) 3815 : Track(thread, client, streamType, sampleRate, format, channelMask, 3816 frameCount, sharedBuffer, sessionId), 3817 mTimedSilenceBuffer(NULL), 3818 mTimedSilenceBufferSize(0), 3819 mTimedAudioOutputOnTime(false), 3820 mMediaTimeTransformValid(false) 3821{ 3822 LocalClock lc; 3823 mLocalTimeFreq = lc.getLocalFreq(); 3824 3825 mLocalTimeToSampleTransform.a_zero = 0; 3826 mLocalTimeToSampleTransform.b_zero = 0; 3827 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3828 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3829 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3830 &mLocalTimeToSampleTransform.a_to_b_denom); 3831} 3832 3833AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3834 mClient->releaseTimedTrack(); 3835 delete [] mTimedSilenceBuffer; 3836} 3837 3838status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3839 size_t size, sp<IMemory>* buffer) { 3840 3841 Mutex::Autolock _l(mTimedBufferQueueLock); 3842 3843 trimTimedBufferQueue_l(); 3844 3845 // lazily initialize the shared memory heap for timed buffers 3846 if (mTimedMemoryDealer == NULL) { 3847 const int kTimedBufferHeapSize = 512 << 10; 3848 3849 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3850 "AudioFlingerTimed"); 3851 if (mTimedMemoryDealer == NULL) 3852 return NO_MEMORY; 3853 } 3854 3855 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3856 if (newBuffer == NULL) { 3857 newBuffer = mTimedMemoryDealer->allocate(size); 3858 if (newBuffer == NULL) 3859 return NO_MEMORY; 3860 } 3861 3862 *buffer = newBuffer; 3863 return NO_ERROR; 3864} 3865 3866// caller must hold mTimedBufferQueueLock 3867void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3868 int64_t mediaTimeNow; 3869 { 3870 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3871 if (!mMediaTimeTransformValid) 3872 return; 3873 3874 int64_t targetTimeNow; 3875 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3876 ? mCCHelper.getCommonTime(&targetTimeNow) 3877 : mCCHelper.getLocalTime(&targetTimeNow); 3878 3879 if (OK != res) 3880 return; 3881 3882 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3883 &mediaTimeNow)) { 3884 return; 3885 } 3886 } 3887 3888 size_t trimIndex; 3889 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3890 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3891 break; 3892 } 3893 3894 if (trimIndex) { 3895 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3896 } 3897} 3898 3899status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3900 const sp<IMemory>& buffer, int64_t pts) { 3901 3902 { 3903 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3904 if (!mMediaTimeTransformValid) 3905 return INVALID_OPERATION; 3906 } 3907 3908 Mutex::Autolock _l(mTimedBufferQueueLock); 3909 3910 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3911 3912 return NO_ERROR; 3913} 3914 3915status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3916 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3917 3918 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3919 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3920 target); 3921 3922 if (!(target == TimedAudioTrack::LOCAL_TIME || 3923 target == TimedAudioTrack::COMMON_TIME)) { 3924 return BAD_VALUE; 3925 } 3926 3927 Mutex::Autolock lock(mMediaTimeTransformLock); 3928 mMediaTimeTransform = xform; 3929 mMediaTimeTransformTarget = target; 3930 mMediaTimeTransformValid = true; 3931 3932 return NO_ERROR; 3933} 3934 3935#define min(a, b) ((a) < (b) ? (a) : (b)) 3936 3937// implementation of getNextBuffer for tracks whose buffers have timestamps 3938status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3939 AudioBufferProvider::Buffer* buffer, int64_t pts) 3940{ 3941 if (pts == AudioBufferProvider::kInvalidPTS) { 3942 buffer->raw = 0; 3943 buffer->frameCount = 0; 3944 return INVALID_OPERATION; 3945 } 3946 3947 Mutex::Autolock _l(mTimedBufferQueueLock); 3948 3949 while (true) { 3950 3951 // if we have no timed buffers, then fail 3952 if (mTimedBufferQueue.isEmpty()) { 3953 buffer->raw = 0; 3954 buffer->frameCount = 0; 3955 return NOT_ENOUGH_DATA; 3956 } 3957 3958 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3959 3960 // calculate the PTS of the head of the timed buffer queue expressed in 3961 // local time 3962 int64_t headLocalPTS; 3963 { 3964 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3965 3966 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 3967 3968 if (mMediaTimeTransform.a_to_b_denom == 0) { 3969 // the transform represents a pause, so yield silence 3970 timedYieldSilence(buffer->frameCount, buffer); 3971 return NO_ERROR; 3972 } 3973 3974 int64_t transformedPTS; 3975 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3976 &transformedPTS)) { 3977 // the transform failed. this shouldn't happen, but if it does 3978 // then just drop this buffer 3979 ALOGW("timedGetNextBuffer transform failed"); 3980 buffer->raw = 0; 3981 buffer->frameCount = 0; 3982 mTimedBufferQueue.removeAt(0); 3983 return NO_ERROR; 3984 } 3985 3986 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3987 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3988 &headLocalPTS)) { 3989 buffer->raw = 0; 3990 buffer->frameCount = 0; 3991 return INVALID_OPERATION; 3992 } 3993 } else { 3994 headLocalPTS = transformedPTS; 3995 } 3996 } 3997 3998 // adjust the head buffer's PTS to reflect the portion of the head buffer 3999 // that has already been consumed 4000 int64_t effectivePTS = headLocalPTS + 4001 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4002 4003 // Calculate the delta in samples between the head of the input buffer 4004 // queue and the start of the next output buffer that will be written. 4005 // If the transformation fails because of over or underflow, it means 4006 // that the sample's position in the output stream is so far out of 4007 // whack that it should just be dropped. 4008 int64_t sampleDelta; 4009 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4010 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4011 mTimedBufferQueue.removeAt(0); 4012 continue; 4013 } 4014 if (!mLocalTimeToSampleTransform.doForwardTransform( 4015 (effectivePTS - pts) << 32, &sampleDelta)) { 4016 ALOGV("*** too late during sample rate transform: dropped buffer"); 4017 mTimedBufferQueue.removeAt(0); 4018 continue; 4019 } 4020 4021 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4022 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4023 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4024 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4025 4026 // if the delta between the ideal placement for the next input sample and 4027 // the current output position is within this threshold, then we will 4028 // concatenate the next input samples to the previous output 4029 const int64_t kSampleContinuityThreshold = 4030 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4031 4032 // if this is the first buffer of audio that we're emitting from this track 4033 // then it should be almost exactly on time. 4034 const int64_t kSampleStartupThreshold = 1LL << 32; 4035 4036 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4037 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4038 // the next input is close enough to being on time, so concatenate it 4039 // with the last output 4040 timedYieldSamples(buffer); 4041 4042 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4043 return NO_ERROR; 4044 } else if (sampleDelta > 0) { 4045 // the gap between the current output position and the proper start of 4046 // the next input sample is too big, so fill it with silence 4047 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4048 4049 timedYieldSilence(framesUntilNextInput, buffer); 4050 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4051 return NO_ERROR; 4052 } else { 4053 // the next input sample is late 4054 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4055 size_t onTimeSamplePosition = 4056 head.position() + lateFrames * mCblk->frameSize; 4057 4058 if (onTimeSamplePosition > head.buffer()->size()) { 4059 // all the remaining samples in the head are too late, so 4060 // drop it and move on 4061 ALOGV("*** too late: dropped buffer"); 4062 mTimedBufferQueue.removeAt(0); 4063 continue; 4064 } else { 4065 // skip over the late samples 4066 head.setPosition(onTimeSamplePosition); 4067 4068 // yield the available samples 4069 timedYieldSamples(buffer); 4070 4071 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4072 return NO_ERROR; 4073 } 4074 } 4075 } 4076} 4077 4078// Yield samples from the timed buffer queue head up to the given output 4079// buffer's capacity. 4080// 4081// Caller must hold mTimedBufferQueueLock 4082void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4083 AudioBufferProvider::Buffer* buffer) { 4084 4085 const TimedBuffer& head = mTimedBufferQueue[0]; 4086 4087 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4088 head.position()); 4089 4090 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4091 mCblk->frameSize); 4092 size_t framesRequested = buffer->frameCount; 4093 buffer->frameCount = min(framesLeftInHead, framesRequested); 4094 4095 mTimedAudioOutputOnTime = true; 4096} 4097 4098// Yield samples of silence up to the given output buffer's capacity 4099// 4100// Caller must hold mTimedBufferQueueLock 4101void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4102 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4103 4104 // lazily allocate a buffer filled with silence 4105 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4106 delete [] mTimedSilenceBuffer; 4107 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4108 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4109 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4110 } 4111 4112 buffer->raw = mTimedSilenceBuffer; 4113 size_t framesRequested = buffer->frameCount; 4114 buffer->frameCount = min(numFrames, framesRequested); 4115 4116 mTimedAudioOutputOnTime = false; 4117} 4118 4119// AudioBufferProvider interface 4120void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4121 AudioBufferProvider::Buffer* buffer) { 4122 4123 Mutex::Autolock _l(mTimedBufferQueueLock); 4124 4125 // If the buffer which was just released is part of the buffer at the head 4126 // of the queue, be sure to update the amt of the buffer which has been 4127 // consumed. If the buffer being returned is not part of the head of the 4128 // queue, its either because the buffer is part of the silence buffer, or 4129 // because the head of the timed queue was trimmed after the mixer called 4130 // getNextBuffer but before the mixer called releaseBuffer. 4131 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4132 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4133 4134 void* start = head.buffer()->pointer(); 4135 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4136 4137 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4138 head.setPosition(head.position() + 4139 (buffer->frameCount * mCblk->frameSize)); 4140 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4141 mTimedBufferQueue.removeAt(0); 4142 } 4143 } 4144 } 4145 4146 buffer->raw = 0; 4147 buffer->frameCount = 0; 4148} 4149 4150uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4151 Mutex::Autolock _l(mTimedBufferQueueLock); 4152 4153 uint32_t frames = 0; 4154 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4155 const TimedBuffer& tb = mTimedBufferQueue[i]; 4156 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4157 } 4158 4159 return frames; 4160} 4161 4162AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4163 : mPTS(0), mPosition(0) {} 4164 4165AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4166 const sp<IMemory>& buffer, int64_t pts) 4167 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4168 4169// ---------------------------------------------------------------------------- 4170 4171// RecordTrack constructor must be called with AudioFlinger::mLock held 4172AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4173 RecordThread *thread, 4174 const sp<Client>& client, 4175 uint32_t sampleRate, 4176 audio_format_t format, 4177 uint32_t channelMask, 4178 int frameCount, 4179 int sessionId) 4180 : TrackBase(thread, client, sampleRate, format, 4181 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4182 mOverflow(false) 4183{ 4184 if (mCblk != NULL) { 4185 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4186 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4187 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4188 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4189 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4190 } else { 4191 mCblk->frameSize = sizeof(int8_t); 4192 } 4193 } 4194} 4195 4196AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4197{ 4198 sp<ThreadBase> thread = mThread.promote(); 4199 if (thread != 0) { 4200 AudioSystem::releaseInput(thread->id()); 4201 } 4202} 4203 4204// AudioBufferProvider interface 4205status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4206{ 4207 audio_track_cblk_t* cblk = this->cblk(); 4208 uint32_t framesAvail; 4209 uint32_t framesReq = buffer->frameCount; 4210 4211 // Check if last stepServer failed, try to step now 4212 if (mStepServerFailed) { 4213 if (!step()) goto getNextBuffer_exit; 4214 ALOGV("stepServer recovered"); 4215 mStepServerFailed = false; 4216 } 4217 4218 framesAvail = cblk->framesAvailable_l(); 4219 4220 if (CC_LIKELY(framesAvail)) { 4221 uint32_t s = cblk->server; 4222 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4223 4224 if (framesReq > framesAvail) { 4225 framesReq = framesAvail; 4226 } 4227 if (s + framesReq > bufferEnd) { 4228 framesReq = bufferEnd - s; 4229 } 4230 4231 buffer->raw = getBuffer(s, framesReq); 4232 if (buffer->raw == NULL) goto getNextBuffer_exit; 4233 4234 buffer->frameCount = framesReq; 4235 return NO_ERROR; 4236 } 4237 4238getNextBuffer_exit: 4239 buffer->raw = NULL; 4240 buffer->frameCount = 0; 4241 return NOT_ENOUGH_DATA; 4242} 4243 4244status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4245{ 4246 sp<ThreadBase> thread = mThread.promote(); 4247 if (thread != 0) { 4248 RecordThread *recordThread = (RecordThread *)thread.get(); 4249 return recordThread->start(this, tid); 4250 } else { 4251 return BAD_VALUE; 4252 } 4253} 4254 4255void AudioFlinger::RecordThread::RecordTrack::stop() 4256{ 4257 sp<ThreadBase> thread = mThread.promote(); 4258 if (thread != 0) { 4259 RecordThread *recordThread = (RecordThread *)thread.get(); 4260 recordThread->stop(this); 4261 TrackBase::reset(); 4262 // Force overrun condition to avoid false overrun callback until first data is 4263 // read from buffer 4264 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4265 } 4266} 4267 4268void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4269{ 4270 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4271 (mClient == 0) ? getpid_cached : mClient->pid(), 4272 mFormat, 4273 mChannelMask, 4274 mSessionId, 4275 mFrameCount, 4276 mState, 4277 mCblk->sampleRate, 4278 mCblk->server, 4279 mCblk->user); 4280} 4281 4282 4283// ---------------------------------------------------------------------------- 4284 4285AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4286 PlaybackThread *playbackThread, 4287 DuplicatingThread *sourceThread, 4288 uint32_t sampleRate, 4289 audio_format_t format, 4290 uint32_t channelMask, 4291 int frameCount) 4292 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4293 mActive(false), mSourceThread(sourceThread) 4294{ 4295 4296 if (mCblk != NULL) { 4297 mCblk->flags |= CBLK_DIRECTION_OUT; 4298 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4299 mOutBuffer.frameCount = 0; 4300 playbackThread->mTracks.add(this); 4301 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4302 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4303 mCblk, mBuffer, mCblk->buffers, 4304 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4305 } else { 4306 ALOGW("Error creating output track on thread %p", playbackThread); 4307 } 4308} 4309 4310AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4311{ 4312 clearBufferQueue(); 4313} 4314 4315status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4316{ 4317 status_t status = Track::start(tid); 4318 if (status != NO_ERROR) { 4319 return status; 4320 } 4321 4322 mActive = true; 4323 mRetryCount = 127; 4324 return status; 4325} 4326 4327void AudioFlinger::PlaybackThread::OutputTrack::stop() 4328{ 4329 Track::stop(); 4330 clearBufferQueue(); 4331 mOutBuffer.frameCount = 0; 4332 mActive = false; 4333} 4334 4335bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4336{ 4337 Buffer *pInBuffer; 4338 Buffer inBuffer; 4339 uint32_t channelCount = mChannelCount; 4340 bool outputBufferFull = false; 4341 inBuffer.frameCount = frames; 4342 inBuffer.i16 = data; 4343 4344 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4345 4346 if (!mActive && frames != 0) { 4347 start(0); 4348 sp<ThreadBase> thread = mThread.promote(); 4349 if (thread != 0) { 4350 MixerThread *mixerThread = (MixerThread *)thread.get(); 4351 if (mCblk->frameCount > frames){ 4352 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4353 uint32_t startFrames = (mCblk->frameCount - frames); 4354 pInBuffer = new Buffer; 4355 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4356 pInBuffer->frameCount = startFrames; 4357 pInBuffer->i16 = pInBuffer->mBuffer; 4358 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4359 mBufferQueue.add(pInBuffer); 4360 } else { 4361 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4362 } 4363 } 4364 } 4365 } 4366 4367 while (waitTimeLeftMs) { 4368 // First write pending buffers, then new data 4369 if (mBufferQueue.size()) { 4370 pInBuffer = mBufferQueue.itemAt(0); 4371 } else { 4372 pInBuffer = &inBuffer; 4373 } 4374 4375 if (pInBuffer->frameCount == 0) { 4376 break; 4377 } 4378 4379 if (mOutBuffer.frameCount == 0) { 4380 mOutBuffer.frameCount = pInBuffer->frameCount; 4381 nsecs_t startTime = systemTime(); 4382 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4383 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4384 outputBufferFull = true; 4385 break; 4386 } 4387 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4388 if (waitTimeLeftMs >= waitTimeMs) { 4389 waitTimeLeftMs -= waitTimeMs; 4390 } else { 4391 waitTimeLeftMs = 0; 4392 } 4393 } 4394 4395 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4396 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4397 mCblk->stepUser(outFrames); 4398 pInBuffer->frameCount -= outFrames; 4399 pInBuffer->i16 += outFrames * channelCount; 4400 mOutBuffer.frameCount -= outFrames; 4401 mOutBuffer.i16 += outFrames * channelCount; 4402 4403 if (pInBuffer->frameCount == 0) { 4404 if (mBufferQueue.size()) { 4405 mBufferQueue.removeAt(0); 4406 delete [] pInBuffer->mBuffer; 4407 delete pInBuffer; 4408 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4409 } else { 4410 break; 4411 } 4412 } 4413 } 4414 4415 // If we could not write all frames, allocate a buffer and queue it for next time. 4416 if (inBuffer.frameCount) { 4417 sp<ThreadBase> thread = mThread.promote(); 4418 if (thread != 0 && !thread->standby()) { 4419 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4420 pInBuffer = new Buffer; 4421 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4422 pInBuffer->frameCount = inBuffer.frameCount; 4423 pInBuffer->i16 = pInBuffer->mBuffer; 4424 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4425 mBufferQueue.add(pInBuffer); 4426 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4427 } else { 4428 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4429 } 4430 } 4431 } 4432 4433 // Calling write() with a 0 length buffer, means that no more data will be written: 4434 // If no more buffers are pending, fill output track buffer to make sure it is started 4435 // by output mixer. 4436 if (frames == 0 && mBufferQueue.size() == 0) { 4437 if (mCblk->user < mCblk->frameCount) { 4438 frames = mCblk->frameCount - mCblk->user; 4439 pInBuffer = new Buffer; 4440 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4441 pInBuffer->frameCount = frames; 4442 pInBuffer->i16 = pInBuffer->mBuffer; 4443 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4444 mBufferQueue.add(pInBuffer); 4445 } else if (mActive) { 4446 stop(); 4447 } 4448 } 4449 4450 return outputBufferFull; 4451} 4452 4453status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4454{ 4455 int active; 4456 status_t result; 4457 audio_track_cblk_t* cblk = mCblk; 4458 uint32_t framesReq = buffer->frameCount; 4459 4460// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4461 buffer->frameCount = 0; 4462 4463 uint32_t framesAvail = cblk->framesAvailable(); 4464 4465 4466 if (framesAvail == 0) { 4467 Mutex::Autolock _l(cblk->lock); 4468 goto start_loop_here; 4469 while (framesAvail == 0) { 4470 active = mActive; 4471 if (CC_UNLIKELY(!active)) { 4472 ALOGV("Not active and NO_MORE_BUFFERS"); 4473 return NO_MORE_BUFFERS; 4474 } 4475 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4476 if (result != NO_ERROR) { 4477 return NO_MORE_BUFFERS; 4478 } 4479 // read the server count again 4480 start_loop_here: 4481 framesAvail = cblk->framesAvailable_l(); 4482 } 4483 } 4484 4485// if (framesAvail < framesReq) { 4486// return NO_MORE_BUFFERS; 4487// } 4488 4489 if (framesReq > framesAvail) { 4490 framesReq = framesAvail; 4491 } 4492 4493 uint32_t u = cblk->user; 4494 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4495 4496 if (u + framesReq > bufferEnd) { 4497 framesReq = bufferEnd - u; 4498 } 4499 4500 buffer->frameCount = framesReq; 4501 buffer->raw = (void *)cblk->buffer(u); 4502 return NO_ERROR; 4503} 4504 4505 4506void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4507{ 4508 size_t size = mBufferQueue.size(); 4509 4510 for (size_t i = 0; i < size; i++) { 4511 Buffer *pBuffer = mBufferQueue.itemAt(i); 4512 delete [] pBuffer->mBuffer; 4513 delete pBuffer; 4514 } 4515 mBufferQueue.clear(); 4516} 4517 4518// ---------------------------------------------------------------------------- 4519 4520AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4521 : RefBase(), 4522 mAudioFlinger(audioFlinger), 4523 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4524 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4525 mPid(pid), 4526 mTimedTrackCount(0) 4527{ 4528 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4529} 4530 4531// Client destructor must be called with AudioFlinger::mLock held 4532AudioFlinger::Client::~Client() 4533{ 4534 mAudioFlinger->removeClient_l(mPid); 4535} 4536 4537sp<MemoryDealer> AudioFlinger::Client::heap() const 4538{ 4539 return mMemoryDealer; 4540} 4541 4542// Reserve one of the limited slots for a timed audio track associated 4543// with this client 4544bool AudioFlinger::Client::reserveTimedTrack() 4545{ 4546 const int kMaxTimedTracksPerClient = 4; 4547 4548 Mutex::Autolock _l(mTimedTrackLock); 4549 4550 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4551 ALOGW("can not create timed track - pid %d has exceeded the limit", 4552 mPid); 4553 return false; 4554 } 4555 4556 mTimedTrackCount++; 4557 return true; 4558} 4559 4560// Release a slot for a timed audio track 4561void AudioFlinger::Client::releaseTimedTrack() 4562{ 4563 Mutex::Autolock _l(mTimedTrackLock); 4564 mTimedTrackCount--; 4565} 4566 4567// ---------------------------------------------------------------------------- 4568 4569AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4570 const sp<IAudioFlingerClient>& client, 4571 pid_t pid) 4572 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4573{ 4574} 4575 4576AudioFlinger::NotificationClient::~NotificationClient() 4577{ 4578} 4579 4580void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4581{ 4582 sp<NotificationClient> keep(this); 4583 mAudioFlinger->removeNotificationClient(mPid); 4584} 4585 4586// ---------------------------------------------------------------------------- 4587 4588AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4589 : BnAudioTrack(), 4590 mTrack(track) 4591{ 4592} 4593 4594AudioFlinger::TrackHandle::~TrackHandle() { 4595 // just stop the track on deletion, associated resources 4596 // will be freed from the main thread once all pending buffers have 4597 // been played. Unless it's not in the active track list, in which 4598 // case we free everything now... 4599 mTrack->destroy(); 4600} 4601 4602sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4603 return mTrack->getCblk(); 4604} 4605 4606status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4607 return mTrack->start(tid); 4608} 4609 4610void AudioFlinger::TrackHandle::stop() { 4611 mTrack->stop(); 4612} 4613 4614void AudioFlinger::TrackHandle::flush() { 4615 mTrack->flush(); 4616} 4617 4618void AudioFlinger::TrackHandle::mute(bool e) { 4619 mTrack->mute(e); 4620} 4621 4622void AudioFlinger::TrackHandle::pause() { 4623 mTrack->pause(); 4624} 4625 4626status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4627{ 4628 return mTrack->attachAuxEffect(EffectId); 4629} 4630 4631status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4632 sp<IMemory>* buffer) { 4633 if (!mTrack->isTimedTrack()) 4634 return INVALID_OPERATION; 4635 4636 PlaybackThread::TimedTrack* tt = 4637 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4638 return tt->allocateTimedBuffer(size, buffer); 4639} 4640 4641status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4642 int64_t pts) { 4643 if (!mTrack->isTimedTrack()) 4644 return INVALID_OPERATION; 4645 4646 PlaybackThread::TimedTrack* tt = 4647 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4648 return tt->queueTimedBuffer(buffer, pts); 4649} 4650 4651status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4652 const LinearTransform& xform, int target) { 4653 4654 if (!mTrack->isTimedTrack()) 4655 return INVALID_OPERATION; 4656 4657 PlaybackThread::TimedTrack* tt = 4658 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4659 return tt->setMediaTimeTransform( 4660 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4661} 4662 4663status_t AudioFlinger::TrackHandle::onTransact( 4664 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4665{ 4666 return BnAudioTrack::onTransact(code, data, reply, flags); 4667} 4668 4669// ---------------------------------------------------------------------------- 4670 4671sp<IAudioRecord> AudioFlinger::openRecord( 4672 pid_t pid, 4673 audio_io_handle_t input, 4674 uint32_t sampleRate, 4675 audio_format_t format, 4676 uint32_t channelMask, 4677 int frameCount, 4678 IAudioFlinger::track_flags_t flags, 4679 int *sessionId, 4680 status_t *status) 4681{ 4682 sp<RecordThread::RecordTrack> recordTrack; 4683 sp<RecordHandle> recordHandle; 4684 sp<Client> client; 4685 status_t lStatus; 4686 RecordThread *thread; 4687 size_t inFrameCount; 4688 int lSessionId; 4689 4690 // check calling permissions 4691 if (!recordingAllowed()) { 4692 lStatus = PERMISSION_DENIED; 4693 goto Exit; 4694 } 4695 4696 // add client to list 4697 { // scope for mLock 4698 Mutex::Autolock _l(mLock); 4699 thread = checkRecordThread_l(input); 4700 if (thread == NULL) { 4701 lStatus = BAD_VALUE; 4702 goto Exit; 4703 } 4704 4705 client = registerPid_l(pid); 4706 4707 // If no audio session id is provided, create one here 4708 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4709 lSessionId = *sessionId; 4710 } else { 4711 lSessionId = nextUniqueId(); 4712 if (sessionId != NULL) { 4713 *sessionId = lSessionId; 4714 } 4715 } 4716 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4717 recordTrack = thread->createRecordTrack_l(client, 4718 sampleRate, 4719 format, 4720 channelMask, 4721 frameCount, 4722 lSessionId, 4723 &lStatus); 4724 } 4725 if (lStatus != NO_ERROR) { 4726 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4727 // destructor is called by the TrackBase destructor with mLock held 4728 client.clear(); 4729 recordTrack.clear(); 4730 goto Exit; 4731 } 4732 4733 // return to handle to client 4734 recordHandle = new RecordHandle(recordTrack); 4735 lStatus = NO_ERROR; 4736 4737Exit: 4738 if (status) { 4739 *status = lStatus; 4740 } 4741 return recordHandle; 4742} 4743 4744// ---------------------------------------------------------------------------- 4745 4746AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4747 : BnAudioRecord(), 4748 mRecordTrack(recordTrack) 4749{ 4750} 4751 4752AudioFlinger::RecordHandle::~RecordHandle() { 4753 stop(); 4754} 4755 4756sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4757 return mRecordTrack->getCblk(); 4758} 4759 4760status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4761 ALOGV("RecordHandle::start()"); 4762 return mRecordTrack->start(tid); 4763} 4764 4765void AudioFlinger::RecordHandle::stop() { 4766 ALOGV("RecordHandle::stop()"); 4767 mRecordTrack->stop(); 4768} 4769 4770status_t AudioFlinger::RecordHandle::onTransact( 4771 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4772{ 4773 return BnAudioRecord::onTransact(code, data, reply, flags); 4774} 4775 4776// ---------------------------------------------------------------------------- 4777 4778AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4779 AudioStreamIn *input, 4780 uint32_t sampleRate, 4781 uint32_t channels, 4782 audio_io_handle_t id, 4783 uint32_t device) : 4784 ThreadBase(audioFlinger, id, device, RECORD), 4785 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4786 // mRsmpInIndex and mInputBytes set by readInputParameters() 4787 mReqChannelCount(popcount(channels)), 4788 mReqSampleRate(sampleRate) 4789 // mBytesRead is only meaningful while active, and so is cleared in start() 4790 // (but might be better to also clear here for dump?) 4791{ 4792 snprintf(mName, kNameLength, "AudioIn_%X", id); 4793 4794 readInputParameters(); 4795} 4796 4797 4798AudioFlinger::RecordThread::~RecordThread() 4799{ 4800 delete[] mRsmpInBuffer; 4801 delete mResampler; 4802 delete[] mRsmpOutBuffer; 4803} 4804 4805void AudioFlinger::RecordThread::onFirstRef() 4806{ 4807 run(mName, PRIORITY_URGENT_AUDIO); 4808} 4809 4810status_t AudioFlinger::RecordThread::readyToRun() 4811{ 4812 status_t status = initCheck(); 4813 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4814 return status; 4815} 4816 4817bool AudioFlinger::RecordThread::threadLoop() 4818{ 4819 AudioBufferProvider::Buffer buffer; 4820 sp<RecordTrack> activeTrack; 4821 Vector< sp<EffectChain> > effectChains; 4822 4823 nsecs_t lastWarning = 0; 4824 4825 acquireWakeLock(); 4826 4827 // start recording 4828 while (!exitPending()) { 4829 4830 processConfigEvents(); 4831 4832 { // scope for mLock 4833 Mutex::Autolock _l(mLock); 4834 checkForNewParameters_l(); 4835 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4836 if (!mStandby) { 4837 mInput->stream->common.standby(&mInput->stream->common); 4838 mStandby = true; 4839 } 4840 4841 if (exitPending()) break; 4842 4843 releaseWakeLock_l(); 4844 ALOGV("RecordThread: loop stopping"); 4845 // go to sleep 4846 mWaitWorkCV.wait(mLock); 4847 ALOGV("RecordThread: loop starting"); 4848 acquireWakeLock_l(); 4849 continue; 4850 } 4851 if (mActiveTrack != 0) { 4852 if (mActiveTrack->mState == TrackBase::PAUSING) { 4853 if (!mStandby) { 4854 mInput->stream->common.standby(&mInput->stream->common); 4855 mStandby = true; 4856 } 4857 mActiveTrack.clear(); 4858 mStartStopCond.broadcast(); 4859 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4860 if (mReqChannelCount != mActiveTrack->channelCount()) { 4861 mActiveTrack.clear(); 4862 mStartStopCond.broadcast(); 4863 } else if (mBytesRead != 0) { 4864 // record start succeeds only if first read from audio input 4865 // succeeds 4866 if (mBytesRead > 0) { 4867 mActiveTrack->mState = TrackBase::ACTIVE; 4868 } else { 4869 mActiveTrack.clear(); 4870 } 4871 mStartStopCond.broadcast(); 4872 } 4873 mStandby = false; 4874 } 4875 } 4876 lockEffectChains_l(effectChains); 4877 } 4878 4879 if (mActiveTrack != 0) { 4880 if (mActiveTrack->mState != TrackBase::ACTIVE && 4881 mActiveTrack->mState != TrackBase::RESUMING) { 4882 unlockEffectChains(effectChains); 4883 usleep(kRecordThreadSleepUs); 4884 continue; 4885 } 4886 for (size_t i = 0; i < effectChains.size(); i ++) { 4887 effectChains[i]->process_l(); 4888 } 4889 4890 buffer.frameCount = mFrameCount; 4891 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4892 size_t framesOut = buffer.frameCount; 4893 if (mResampler == NULL) { 4894 // no resampling 4895 while (framesOut) { 4896 size_t framesIn = mFrameCount - mRsmpInIndex; 4897 if (framesIn) { 4898 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4899 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4900 if (framesIn > framesOut) 4901 framesIn = framesOut; 4902 mRsmpInIndex += framesIn; 4903 framesOut -= framesIn; 4904 if ((int)mChannelCount == mReqChannelCount || 4905 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4906 memcpy(dst, src, framesIn * mFrameSize); 4907 } else { 4908 int16_t *src16 = (int16_t *)src; 4909 int16_t *dst16 = (int16_t *)dst; 4910 if (mChannelCount == 1) { 4911 while (framesIn--) { 4912 *dst16++ = *src16; 4913 *dst16++ = *src16++; 4914 } 4915 } else { 4916 while (framesIn--) { 4917 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4918 src16 += 2; 4919 } 4920 } 4921 } 4922 } 4923 if (framesOut && mFrameCount == mRsmpInIndex) { 4924 if (framesOut == mFrameCount && 4925 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4926 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4927 framesOut = 0; 4928 } else { 4929 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4930 mRsmpInIndex = 0; 4931 } 4932 if (mBytesRead < 0) { 4933 ALOGE("Error reading audio input"); 4934 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4935 // Force input into standby so that it tries to 4936 // recover at next read attempt 4937 mInput->stream->common.standby(&mInput->stream->common); 4938 usleep(kRecordThreadSleepUs); 4939 } 4940 mRsmpInIndex = mFrameCount; 4941 framesOut = 0; 4942 buffer.frameCount = 0; 4943 } 4944 } 4945 } 4946 } else { 4947 // resampling 4948 4949 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4950 // alter output frame count as if we were expecting stereo samples 4951 if (mChannelCount == 1 && mReqChannelCount == 1) { 4952 framesOut >>= 1; 4953 } 4954 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4955 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4956 // are 32 bit aligned which should be always true. 4957 if (mChannelCount == 2 && mReqChannelCount == 1) { 4958 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4959 // the resampler always outputs stereo samples: do post stereo to mono conversion 4960 int16_t *src = (int16_t *)mRsmpOutBuffer; 4961 int16_t *dst = buffer.i16; 4962 while (framesOut--) { 4963 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4964 src += 2; 4965 } 4966 } else { 4967 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4968 } 4969 4970 } 4971 mActiveTrack->releaseBuffer(&buffer); 4972 mActiveTrack->overflow(); 4973 } 4974 // client isn't retrieving buffers fast enough 4975 else { 4976 if (!mActiveTrack->setOverflow()) { 4977 nsecs_t now = systemTime(); 4978 if ((now - lastWarning) > kWarningThrottleNs) { 4979 ALOGW("RecordThread: buffer overflow"); 4980 lastWarning = now; 4981 } 4982 } 4983 // Release the processor for a while before asking for a new buffer. 4984 // This will give the application more chance to read from the buffer and 4985 // clear the overflow. 4986 usleep(kRecordThreadSleepUs); 4987 } 4988 } 4989 // enable changes in effect chain 4990 unlockEffectChains(effectChains); 4991 effectChains.clear(); 4992 } 4993 4994 if (!mStandby) { 4995 mInput->stream->common.standby(&mInput->stream->common); 4996 } 4997 mActiveTrack.clear(); 4998 4999 mStartStopCond.broadcast(); 5000 5001 releaseWakeLock(); 5002 5003 ALOGV("RecordThread %p exiting", this); 5004 return false; 5005} 5006 5007 5008sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5009 const sp<AudioFlinger::Client>& client, 5010 uint32_t sampleRate, 5011 audio_format_t format, 5012 int channelMask, 5013 int frameCount, 5014 int sessionId, 5015 status_t *status) 5016{ 5017 sp<RecordTrack> track; 5018 status_t lStatus; 5019 5020 lStatus = initCheck(); 5021 if (lStatus != NO_ERROR) { 5022 ALOGE("Audio driver not initialized."); 5023 goto Exit; 5024 } 5025 5026 { // scope for mLock 5027 Mutex::Autolock _l(mLock); 5028 5029 track = new RecordTrack(this, client, sampleRate, 5030 format, channelMask, frameCount, sessionId); 5031 5032 if (track->getCblk() == 0) { 5033 lStatus = NO_MEMORY; 5034 goto Exit; 5035 } 5036 5037 mTrack = track.get(); 5038 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5039 bool suspend = audio_is_bluetooth_sco_device( 5040 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5041 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5042 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5043 } 5044 lStatus = NO_ERROR; 5045 5046Exit: 5047 if (status) { 5048 *status = lStatus; 5049 } 5050 return track; 5051} 5052 5053status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5054{ 5055 ALOGV("RecordThread::start tid=%d", tid); 5056 sp<ThreadBase> strongMe = this; 5057 status_t status = NO_ERROR; 5058 { 5059 AutoMutex lock(mLock); 5060 if (mActiveTrack != 0) { 5061 if (recordTrack != mActiveTrack.get()) { 5062 status = -EBUSY; 5063 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5064 mActiveTrack->mState = TrackBase::ACTIVE; 5065 } 5066 return status; 5067 } 5068 5069 recordTrack->mState = TrackBase::IDLE; 5070 mActiveTrack = recordTrack; 5071 mLock.unlock(); 5072 status_t status = AudioSystem::startInput(mId); 5073 mLock.lock(); 5074 if (status != NO_ERROR) { 5075 mActiveTrack.clear(); 5076 return status; 5077 } 5078 mRsmpInIndex = mFrameCount; 5079 mBytesRead = 0; 5080 if (mResampler != NULL) { 5081 mResampler->reset(); 5082 } 5083 mActiveTrack->mState = TrackBase::RESUMING; 5084 // signal thread to start 5085 ALOGV("Signal record thread"); 5086 mWaitWorkCV.signal(); 5087 // do not wait for mStartStopCond if exiting 5088 if (exitPending()) { 5089 mActiveTrack.clear(); 5090 status = INVALID_OPERATION; 5091 goto startError; 5092 } 5093 mStartStopCond.wait(mLock); 5094 if (mActiveTrack == 0) { 5095 ALOGV("Record failed to start"); 5096 status = BAD_VALUE; 5097 goto startError; 5098 } 5099 ALOGV("Record started OK"); 5100 return status; 5101 } 5102startError: 5103 AudioSystem::stopInput(mId); 5104 return status; 5105} 5106 5107void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5108 ALOGV("RecordThread::stop"); 5109 sp<ThreadBase> strongMe = this; 5110 { 5111 AutoMutex lock(mLock); 5112 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5113 mActiveTrack->mState = TrackBase::PAUSING; 5114 // do not wait for mStartStopCond if exiting 5115 if (exitPending()) { 5116 return; 5117 } 5118 mStartStopCond.wait(mLock); 5119 // if we have been restarted, recordTrack == mActiveTrack.get() here 5120 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5121 mLock.unlock(); 5122 AudioSystem::stopInput(mId); 5123 mLock.lock(); 5124 ALOGV("Record stopped OK"); 5125 } 5126 } 5127 } 5128} 5129 5130status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5131{ 5132 const size_t SIZE = 256; 5133 char buffer[SIZE]; 5134 String8 result; 5135 5136 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5137 result.append(buffer); 5138 5139 if (mActiveTrack != 0) { 5140 result.append("Active Track:\n"); 5141 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5142 mActiveTrack->dump(buffer, SIZE); 5143 result.append(buffer); 5144 5145 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5146 result.append(buffer); 5147 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5148 result.append(buffer); 5149 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5150 result.append(buffer); 5151 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5152 result.append(buffer); 5153 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5154 result.append(buffer); 5155 5156 5157 } else { 5158 result.append("No record client\n"); 5159 } 5160 write(fd, result.string(), result.size()); 5161 5162 dumpBase(fd, args); 5163 dumpEffectChains(fd, args); 5164 5165 return NO_ERROR; 5166} 5167 5168// AudioBufferProvider interface 5169status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5170{ 5171 size_t framesReq = buffer->frameCount; 5172 size_t framesReady = mFrameCount - mRsmpInIndex; 5173 int channelCount; 5174 5175 if (framesReady == 0) { 5176 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5177 if (mBytesRead < 0) { 5178 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5179 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5180 // Force input into standby so that it tries to 5181 // recover at next read attempt 5182 mInput->stream->common.standby(&mInput->stream->common); 5183 usleep(kRecordThreadSleepUs); 5184 } 5185 buffer->raw = NULL; 5186 buffer->frameCount = 0; 5187 return NOT_ENOUGH_DATA; 5188 } 5189 mRsmpInIndex = 0; 5190 framesReady = mFrameCount; 5191 } 5192 5193 if (framesReq > framesReady) { 5194 framesReq = framesReady; 5195 } 5196 5197 if (mChannelCount == 1 && mReqChannelCount == 2) { 5198 channelCount = 1; 5199 } else { 5200 channelCount = 2; 5201 } 5202 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5203 buffer->frameCount = framesReq; 5204 return NO_ERROR; 5205} 5206 5207// AudioBufferProvider interface 5208void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5209{ 5210 mRsmpInIndex += buffer->frameCount; 5211 buffer->frameCount = 0; 5212} 5213 5214bool AudioFlinger::RecordThread::checkForNewParameters_l() 5215{ 5216 bool reconfig = false; 5217 5218 while (!mNewParameters.isEmpty()) { 5219 status_t status = NO_ERROR; 5220 String8 keyValuePair = mNewParameters[0]; 5221 AudioParameter param = AudioParameter(keyValuePair); 5222 int value; 5223 audio_format_t reqFormat = mFormat; 5224 int reqSamplingRate = mReqSampleRate; 5225 int reqChannelCount = mReqChannelCount; 5226 5227 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5228 reqSamplingRate = value; 5229 reconfig = true; 5230 } 5231 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5232 reqFormat = (audio_format_t) value; 5233 reconfig = true; 5234 } 5235 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5236 reqChannelCount = popcount(value); 5237 reconfig = true; 5238 } 5239 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5240 // do not accept frame count changes if tracks are open as the track buffer 5241 // size depends on frame count and correct behavior would not be guaranteed 5242 // if frame count is changed after track creation 5243 if (mActiveTrack != 0) { 5244 status = INVALID_OPERATION; 5245 } else { 5246 reconfig = true; 5247 } 5248 } 5249 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5250 // forward device change to effects that have requested to be 5251 // aware of attached audio device. 5252 for (size_t i = 0; i < mEffectChains.size(); i++) { 5253 mEffectChains[i]->setDevice_l(value); 5254 } 5255 // store input device and output device but do not forward output device to audio HAL. 5256 // Note that status is ignored by the caller for output device 5257 // (see AudioFlinger::setParameters() 5258 if (value & AUDIO_DEVICE_OUT_ALL) { 5259 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5260 status = BAD_VALUE; 5261 } else { 5262 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5263 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5264 if (mTrack != NULL) { 5265 bool suspend = audio_is_bluetooth_sco_device( 5266 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5267 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5268 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5269 } 5270 } 5271 mDevice |= (uint32_t)value; 5272 } 5273 if (status == NO_ERROR) { 5274 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5275 if (status == INVALID_OPERATION) { 5276 mInput->stream->common.standby(&mInput->stream->common); 5277 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5278 keyValuePair.string()); 5279 } 5280 if (reconfig) { 5281 if (status == BAD_VALUE && 5282 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5283 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5284 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5285 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5286 (reqChannelCount <= FCC_2)) { 5287 status = NO_ERROR; 5288 } 5289 if (status == NO_ERROR) { 5290 readInputParameters(); 5291 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5292 } 5293 } 5294 } 5295 5296 mNewParameters.removeAt(0); 5297 5298 mParamStatus = status; 5299 mParamCond.signal(); 5300 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5301 // already timed out waiting for the status and will never signal the condition. 5302 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5303 } 5304 return reconfig; 5305} 5306 5307String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5308{ 5309 char *s; 5310 String8 out_s8 = String8(); 5311 5312 Mutex::Autolock _l(mLock); 5313 if (initCheck() != NO_ERROR) { 5314 return out_s8; 5315 } 5316 5317 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5318 out_s8 = String8(s); 5319 free(s); 5320 return out_s8; 5321} 5322 5323void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5324 AudioSystem::OutputDescriptor desc; 5325 void *param2 = NULL; 5326 5327 switch (event) { 5328 case AudioSystem::INPUT_OPENED: 5329 case AudioSystem::INPUT_CONFIG_CHANGED: 5330 desc.channels = mChannelMask; 5331 desc.samplingRate = mSampleRate; 5332 desc.format = mFormat; 5333 desc.frameCount = mFrameCount; 5334 desc.latency = 0; 5335 param2 = &desc; 5336 break; 5337 5338 case AudioSystem::INPUT_CLOSED: 5339 default: 5340 break; 5341 } 5342 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5343} 5344 5345void AudioFlinger::RecordThread::readInputParameters() 5346{ 5347 delete mRsmpInBuffer; 5348 // mRsmpInBuffer is always assigned a new[] below 5349 delete mRsmpOutBuffer; 5350 mRsmpOutBuffer = NULL; 5351 delete mResampler; 5352 mResampler = NULL; 5353 5354 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5355 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5356 mChannelCount = (uint16_t)popcount(mChannelMask); 5357 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5358 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5359 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5360 mFrameCount = mInputBytes / mFrameSize; 5361 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5362 5363 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5364 { 5365 int channelCount; 5366 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5367 // stereo to mono post process as the resampler always outputs stereo. 5368 if (mChannelCount == 1 && mReqChannelCount == 2) { 5369 channelCount = 1; 5370 } else { 5371 channelCount = 2; 5372 } 5373 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5374 mResampler->setSampleRate(mSampleRate); 5375 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5376 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5377 5378 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5379 if (mChannelCount == 1 && mReqChannelCount == 1) { 5380 mFrameCount >>= 1; 5381 } 5382 5383 } 5384 mRsmpInIndex = mFrameCount; 5385} 5386 5387unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5388{ 5389 Mutex::Autolock _l(mLock); 5390 if (initCheck() != NO_ERROR) { 5391 return 0; 5392 } 5393 5394 return mInput->stream->get_input_frames_lost(mInput->stream); 5395} 5396 5397uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5398{ 5399 Mutex::Autolock _l(mLock); 5400 uint32_t result = 0; 5401 if (getEffectChain_l(sessionId) != 0) { 5402 result = EFFECT_SESSION; 5403 } 5404 5405 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5406 result |= TRACK_SESSION; 5407 } 5408 5409 return result; 5410} 5411 5412AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5413{ 5414 Mutex::Autolock _l(mLock); 5415 return mTrack; 5416} 5417 5418AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5419{ 5420 Mutex::Autolock _l(mLock); 5421 return mInput; 5422} 5423 5424AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5425{ 5426 Mutex::Autolock _l(mLock); 5427 AudioStreamIn *input = mInput; 5428 mInput = NULL; 5429 return input; 5430} 5431 5432// this method must always be called either with ThreadBase mLock held or inside the thread loop 5433audio_stream_t* AudioFlinger::RecordThread::stream() 5434{ 5435 if (mInput == NULL) { 5436 return NULL; 5437 } 5438 return &mInput->stream->common; 5439} 5440 5441 5442// ---------------------------------------------------------------------------- 5443 5444audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5445 uint32_t *pSamplingRate, 5446 audio_format_t *pFormat, 5447 uint32_t *pChannels, 5448 uint32_t *pLatencyMs, 5449 audio_policy_output_flags_t flags) 5450{ 5451 status_t status; 5452 PlaybackThread *thread = NULL; 5453 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5454 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5455 uint32_t channels = pChannels ? *pChannels : 0; 5456 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5457 audio_stream_out_t *outStream; 5458 audio_hw_device_t *outHwDev; 5459 5460 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5461 pDevices ? *pDevices : 0, 5462 samplingRate, 5463 format, 5464 channels, 5465 flags); 5466 5467 if (pDevices == NULL || *pDevices == 0) { 5468 return 0; 5469 } 5470 5471 Mutex::Autolock _l(mLock); 5472 5473 outHwDev = findSuitableHwDev_l(*pDevices); 5474 if (outHwDev == NULL) 5475 return 0; 5476 5477 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5478 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5479 &channels, &samplingRate, &outStream); 5480 mHardwareStatus = AUDIO_HW_IDLE; 5481 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5482 outStream, 5483 samplingRate, 5484 format, 5485 channels, 5486 status); 5487 5488 if (outStream != NULL) { 5489 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5490 audio_io_handle_t id = nextUniqueId(); 5491 5492 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5493 (format != AUDIO_FORMAT_PCM_16_BIT) || 5494 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5495 thread = new DirectOutputThread(this, output, id, *pDevices); 5496 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5497 } else { 5498 thread = new MixerThread(this, output, id, *pDevices); 5499 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5500 } 5501 mPlaybackThreads.add(id, thread); 5502 5503 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5504 if (pFormat != NULL) *pFormat = format; 5505 if (pChannels != NULL) *pChannels = channels; 5506 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5507 5508 // notify client processes of the new output creation 5509 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5510 return id; 5511 } 5512 5513 return 0; 5514} 5515 5516audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5517 audio_io_handle_t output2) 5518{ 5519 Mutex::Autolock _l(mLock); 5520 MixerThread *thread1 = checkMixerThread_l(output1); 5521 MixerThread *thread2 = checkMixerThread_l(output2); 5522 5523 if (thread1 == NULL || thread2 == NULL) { 5524 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5525 return 0; 5526 } 5527 5528 audio_io_handle_t id = nextUniqueId(); 5529 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5530 thread->addOutputTrack(thread2); 5531 mPlaybackThreads.add(id, thread); 5532 // notify client processes of the new output creation 5533 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5534 return id; 5535} 5536 5537status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5538{ 5539 // keep strong reference on the playback thread so that 5540 // it is not destroyed while exit() is executed 5541 sp<PlaybackThread> thread; 5542 { 5543 Mutex::Autolock _l(mLock); 5544 thread = checkPlaybackThread_l(output); 5545 if (thread == NULL) { 5546 return BAD_VALUE; 5547 } 5548 5549 ALOGV("closeOutput() %d", output); 5550 5551 if (thread->type() == ThreadBase::MIXER) { 5552 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5553 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5554 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5555 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5556 } 5557 } 5558 } 5559 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5560 mPlaybackThreads.removeItem(output); 5561 } 5562 thread->exit(); 5563 // The thread entity (active unit of execution) is no longer running here, 5564 // but the ThreadBase container still exists. 5565 5566 if (thread->type() != ThreadBase::DUPLICATING) { 5567 AudioStreamOut *out = thread->clearOutput(); 5568 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5569 // from now on thread->mOutput is NULL 5570 out->hwDev->close_output_stream(out->hwDev, out->stream); 5571 delete out; 5572 } 5573 return NO_ERROR; 5574} 5575 5576status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5577{ 5578 Mutex::Autolock _l(mLock); 5579 PlaybackThread *thread = checkPlaybackThread_l(output); 5580 5581 if (thread == NULL) { 5582 return BAD_VALUE; 5583 } 5584 5585 ALOGV("suspendOutput() %d", output); 5586 thread->suspend(); 5587 5588 return NO_ERROR; 5589} 5590 5591status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5592{ 5593 Mutex::Autolock _l(mLock); 5594 PlaybackThread *thread = checkPlaybackThread_l(output); 5595 5596 if (thread == NULL) { 5597 return BAD_VALUE; 5598 } 5599 5600 ALOGV("restoreOutput() %d", output); 5601 5602 thread->restore(); 5603 5604 return NO_ERROR; 5605} 5606 5607audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5608 uint32_t *pSamplingRate, 5609 audio_format_t *pFormat, 5610 uint32_t *pChannels, 5611 audio_in_acoustics_t acoustics) 5612{ 5613 status_t status; 5614 RecordThread *thread = NULL; 5615 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5616 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5617 uint32_t channels = pChannels ? *pChannels : 0; 5618 uint32_t reqSamplingRate = samplingRate; 5619 audio_format_t reqFormat = format; 5620 uint32_t reqChannels = channels; 5621 audio_stream_in_t *inStream; 5622 audio_hw_device_t *inHwDev; 5623 5624 if (pDevices == NULL || *pDevices == 0) { 5625 return 0; 5626 } 5627 5628 Mutex::Autolock _l(mLock); 5629 5630 inHwDev = findSuitableHwDev_l(*pDevices); 5631 if (inHwDev == NULL) 5632 return 0; 5633 5634 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5635 &channels, &samplingRate, 5636 acoustics, 5637 &inStream); 5638 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5639 inStream, 5640 samplingRate, 5641 format, 5642 channels, 5643 acoustics, 5644 status); 5645 5646 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5647 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5648 // or stereo to mono conversions on 16 bit PCM inputs. 5649 if (inStream == NULL && status == BAD_VALUE && 5650 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5651 (samplingRate <= 2 * reqSamplingRate) && 5652 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5653 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5654 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5655 &channels, &samplingRate, 5656 acoustics, 5657 &inStream); 5658 } 5659 5660 if (inStream != NULL) { 5661 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5662 5663 audio_io_handle_t id = nextUniqueId(); 5664 // Start record thread 5665 // RecorThread require both input and output device indication to forward to audio 5666 // pre processing modules 5667 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5668 thread = new RecordThread(this, 5669 input, 5670 reqSamplingRate, 5671 reqChannels, 5672 id, 5673 device); 5674 mRecordThreads.add(id, thread); 5675 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5676 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5677 if (pFormat != NULL) *pFormat = format; 5678 if (pChannels != NULL) *pChannels = reqChannels; 5679 5680 input->stream->common.standby(&input->stream->common); 5681 5682 // notify client processes of the new input creation 5683 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5684 return id; 5685 } 5686 5687 return 0; 5688} 5689 5690status_t AudioFlinger::closeInput(audio_io_handle_t input) 5691{ 5692 // keep strong reference on the record thread so that 5693 // it is not destroyed while exit() is executed 5694 sp<RecordThread> thread; 5695 { 5696 Mutex::Autolock _l(mLock); 5697 thread = checkRecordThread_l(input); 5698 if (thread == NULL) { 5699 return BAD_VALUE; 5700 } 5701 5702 ALOGV("closeInput() %d", input); 5703 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5704 mRecordThreads.removeItem(input); 5705 } 5706 thread->exit(); 5707 // The thread entity (active unit of execution) is no longer running here, 5708 // but the ThreadBase container still exists. 5709 5710 AudioStreamIn *in = thread->clearInput(); 5711 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5712 // from now on thread->mInput is NULL 5713 in->hwDev->close_input_stream(in->hwDev, in->stream); 5714 delete in; 5715 5716 return NO_ERROR; 5717} 5718 5719status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5720{ 5721 Mutex::Autolock _l(mLock); 5722 MixerThread *dstThread = checkMixerThread_l(output); 5723 if (dstThread == NULL) { 5724 ALOGW("setStreamOutput() bad output id %d", output); 5725 return BAD_VALUE; 5726 } 5727 5728 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5729 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5730 5731 dstThread->setStreamValid(stream, true); 5732 5733 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5734 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5735 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5736 MixerThread *srcThread = (MixerThread *)thread; 5737 srcThread->setStreamValid(stream, false); 5738 srcThread->invalidateTracks(stream); 5739 } 5740 } 5741 5742 return NO_ERROR; 5743} 5744 5745 5746int AudioFlinger::newAudioSessionId() 5747{ 5748 return nextUniqueId(); 5749} 5750 5751void AudioFlinger::acquireAudioSessionId(int audioSession) 5752{ 5753 Mutex::Autolock _l(mLock); 5754 pid_t caller = IPCThreadState::self()->getCallingPid(); 5755 ALOGV("acquiring %d from %d", audioSession, caller); 5756 size_t num = mAudioSessionRefs.size(); 5757 for (size_t i = 0; i< num; i++) { 5758 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5759 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5760 ref->mCnt++; 5761 ALOGV(" incremented refcount to %d", ref->mCnt); 5762 return; 5763 } 5764 } 5765 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5766 ALOGV(" added new entry for %d", audioSession); 5767} 5768 5769void AudioFlinger::releaseAudioSessionId(int audioSession) 5770{ 5771 Mutex::Autolock _l(mLock); 5772 pid_t caller = IPCThreadState::self()->getCallingPid(); 5773 ALOGV("releasing %d from %d", audioSession, caller); 5774 size_t num = mAudioSessionRefs.size(); 5775 for (size_t i = 0; i< num; i++) { 5776 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5777 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5778 ref->mCnt--; 5779 ALOGV(" decremented refcount to %d", ref->mCnt); 5780 if (ref->mCnt == 0) { 5781 mAudioSessionRefs.removeAt(i); 5782 delete ref; 5783 purgeStaleEffects_l(); 5784 } 5785 return; 5786 } 5787 } 5788 ALOGW("session id %d not found for pid %d", audioSession, caller); 5789} 5790 5791void AudioFlinger::purgeStaleEffects_l() { 5792 5793 ALOGV("purging stale effects"); 5794 5795 Vector< sp<EffectChain> > chains; 5796 5797 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5798 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5799 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5800 sp<EffectChain> ec = t->mEffectChains[j]; 5801 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5802 chains.push(ec); 5803 } 5804 } 5805 } 5806 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5807 sp<RecordThread> t = mRecordThreads.valueAt(i); 5808 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5809 sp<EffectChain> ec = t->mEffectChains[j]; 5810 chains.push(ec); 5811 } 5812 } 5813 5814 for (size_t i = 0; i < chains.size(); i++) { 5815 sp<EffectChain> ec = chains[i]; 5816 int sessionid = ec->sessionId(); 5817 sp<ThreadBase> t = ec->mThread.promote(); 5818 if (t == 0) { 5819 continue; 5820 } 5821 size_t numsessionrefs = mAudioSessionRefs.size(); 5822 bool found = false; 5823 for (size_t k = 0; k < numsessionrefs; k++) { 5824 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5825 if (ref->mSessionid == sessionid) { 5826 ALOGV(" session %d still exists for %d with %d refs", 5827 sessionid, ref->mPid, ref->mCnt); 5828 found = true; 5829 break; 5830 } 5831 } 5832 if (!found) { 5833 // remove all effects from the chain 5834 while (ec->mEffects.size()) { 5835 sp<EffectModule> effect = ec->mEffects[0]; 5836 effect->unPin(); 5837 Mutex::Autolock _l (t->mLock); 5838 t->removeEffect_l(effect); 5839 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5840 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5841 if (handle != 0) { 5842 handle->mEffect.clear(); 5843 if (handle->mHasControl && handle->mEnabled) { 5844 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5845 } 5846 } 5847 } 5848 AudioSystem::unregisterEffect(effect->id()); 5849 } 5850 } 5851 } 5852 return; 5853} 5854 5855// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5856AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5857{ 5858 return mPlaybackThreads.valueFor(output).get(); 5859} 5860 5861// checkMixerThread_l() must be called with AudioFlinger::mLock held 5862AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5863{ 5864 PlaybackThread *thread = checkPlaybackThread_l(output); 5865 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5866} 5867 5868// checkRecordThread_l() must be called with AudioFlinger::mLock held 5869AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5870{ 5871 return mRecordThreads.valueFor(input).get(); 5872} 5873 5874uint32_t AudioFlinger::nextUniqueId() 5875{ 5876 return android_atomic_inc(&mNextUniqueId); 5877} 5878 5879AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5880{ 5881 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5882 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5883 AudioStreamOut *output = thread->getOutput(); 5884 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5885 return thread; 5886 } 5887 } 5888 return NULL; 5889} 5890 5891uint32_t AudioFlinger::primaryOutputDevice_l() const 5892{ 5893 PlaybackThread *thread = primaryPlaybackThread_l(); 5894 5895 if (thread == NULL) { 5896 return 0; 5897 } 5898 5899 return thread->device(); 5900} 5901 5902 5903// ---------------------------------------------------------------------------- 5904// Effect management 5905// ---------------------------------------------------------------------------- 5906 5907 5908status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5909{ 5910 Mutex::Autolock _l(mLock); 5911 return EffectQueryNumberEffects(numEffects); 5912} 5913 5914status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5915{ 5916 Mutex::Autolock _l(mLock); 5917 return EffectQueryEffect(index, descriptor); 5918} 5919 5920status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5921 effect_descriptor_t *descriptor) const 5922{ 5923 Mutex::Autolock _l(mLock); 5924 return EffectGetDescriptor(pUuid, descriptor); 5925} 5926 5927 5928sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5929 effect_descriptor_t *pDesc, 5930 const sp<IEffectClient>& effectClient, 5931 int32_t priority, 5932 audio_io_handle_t io, 5933 int sessionId, 5934 status_t *status, 5935 int *id, 5936 int *enabled) 5937{ 5938 status_t lStatus = NO_ERROR; 5939 sp<EffectHandle> handle; 5940 effect_descriptor_t desc; 5941 5942 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5943 pid, effectClient.get(), priority, sessionId, io); 5944 5945 if (pDesc == NULL) { 5946 lStatus = BAD_VALUE; 5947 goto Exit; 5948 } 5949 5950 // check audio settings permission for global effects 5951 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5952 lStatus = PERMISSION_DENIED; 5953 goto Exit; 5954 } 5955 5956 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5957 // that can only be created by audio policy manager (running in same process) 5958 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5959 lStatus = PERMISSION_DENIED; 5960 goto Exit; 5961 } 5962 5963 if (io == 0) { 5964 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5965 // output must be specified by AudioPolicyManager when using session 5966 // AUDIO_SESSION_OUTPUT_STAGE 5967 lStatus = BAD_VALUE; 5968 goto Exit; 5969 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5970 // if the output returned by getOutputForEffect() is removed before we lock the 5971 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5972 // and we will exit safely 5973 io = AudioSystem::getOutputForEffect(&desc); 5974 } 5975 } 5976 5977 { 5978 Mutex::Autolock _l(mLock); 5979 5980 5981 if (!EffectIsNullUuid(&pDesc->uuid)) { 5982 // if uuid is specified, request effect descriptor 5983 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5984 if (lStatus < 0) { 5985 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5986 goto Exit; 5987 } 5988 } else { 5989 // if uuid is not specified, look for an available implementation 5990 // of the required type in effect factory 5991 if (EffectIsNullUuid(&pDesc->type)) { 5992 ALOGW("createEffect() no effect type"); 5993 lStatus = BAD_VALUE; 5994 goto Exit; 5995 } 5996 uint32_t numEffects = 0; 5997 effect_descriptor_t d; 5998 d.flags = 0; // prevent compiler warning 5999 bool found = false; 6000 6001 lStatus = EffectQueryNumberEffects(&numEffects); 6002 if (lStatus < 0) { 6003 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6004 goto Exit; 6005 } 6006 for (uint32_t i = 0; i < numEffects; i++) { 6007 lStatus = EffectQueryEffect(i, &desc); 6008 if (lStatus < 0) { 6009 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6010 continue; 6011 } 6012 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6013 // If matching type found save effect descriptor. If the session is 6014 // 0 and the effect is not auxiliary, continue enumeration in case 6015 // an auxiliary version of this effect type is available 6016 found = true; 6017 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6018 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6019 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6020 break; 6021 } 6022 } 6023 } 6024 if (!found) { 6025 lStatus = BAD_VALUE; 6026 ALOGW("createEffect() effect not found"); 6027 goto Exit; 6028 } 6029 // For same effect type, chose auxiliary version over insert version if 6030 // connect to output mix (Compliance to OpenSL ES) 6031 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6032 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6033 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6034 } 6035 } 6036 6037 // Do not allow auxiliary effects on a session different from 0 (output mix) 6038 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6039 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6040 lStatus = INVALID_OPERATION; 6041 goto Exit; 6042 } 6043 6044 // check recording permission for visualizer 6045 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6046 !recordingAllowed()) { 6047 lStatus = PERMISSION_DENIED; 6048 goto Exit; 6049 } 6050 6051 // return effect descriptor 6052 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6053 6054 // If output is not specified try to find a matching audio session ID in one of the 6055 // output threads. 6056 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6057 // because of code checking output when entering the function. 6058 // Note: io is never 0 when creating an effect on an input 6059 if (io == 0) { 6060 // look for the thread where the specified audio session is present 6061 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6062 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6063 io = mPlaybackThreads.keyAt(i); 6064 break; 6065 } 6066 } 6067 if (io == 0) { 6068 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6069 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6070 io = mRecordThreads.keyAt(i); 6071 break; 6072 } 6073 } 6074 } 6075 // If no output thread contains the requested session ID, default to 6076 // first output. The effect chain will be moved to the correct output 6077 // thread when a track with the same session ID is created 6078 if (io == 0 && mPlaybackThreads.size()) { 6079 io = mPlaybackThreads.keyAt(0); 6080 } 6081 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6082 } 6083 ThreadBase *thread = checkRecordThread_l(io); 6084 if (thread == NULL) { 6085 thread = checkPlaybackThread_l(io); 6086 if (thread == NULL) { 6087 ALOGE("createEffect() unknown output thread"); 6088 lStatus = BAD_VALUE; 6089 goto Exit; 6090 } 6091 } 6092 6093 sp<Client> client = registerPid_l(pid); 6094 6095 // create effect on selected output thread 6096 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6097 &desc, enabled, &lStatus); 6098 if (handle != 0 && id != NULL) { 6099 *id = handle->id(); 6100 } 6101 } 6102 6103Exit: 6104 if (status != NULL) { 6105 *status = lStatus; 6106 } 6107 return handle; 6108} 6109 6110status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6111 audio_io_handle_t dstOutput) 6112{ 6113 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6114 sessionId, srcOutput, dstOutput); 6115 Mutex::Autolock _l(mLock); 6116 if (srcOutput == dstOutput) { 6117 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6118 return NO_ERROR; 6119 } 6120 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6121 if (srcThread == NULL) { 6122 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6123 return BAD_VALUE; 6124 } 6125 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6126 if (dstThread == NULL) { 6127 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6128 return BAD_VALUE; 6129 } 6130 6131 Mutex::Autolock _dl(dstThread->mLock); 6132 Mutex::Autolock _sl(srcThread->mLock); 6133 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6134 6135 return NO_ERROR; 6136} 6137 6138// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6139status_t AudioFlinger::moveEffectChain_l(int sessionId, 6140 AudioFlinger::PlaybackThread *srcThread, 6141 AudioFlinger::PlaybackThread *dstThread, 6142 bool reRegister) 6143{ 6144 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6145 sessionId, srcThread, dstThread); 6146 6147 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6148 if (chain == 0) { 6149 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6150 sessionId, srcThread); 6151 return INVALID_OPERATION; 6152 } 6153 6154 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6155 // so that a new chain is created with correct parameters when first effect is added. This is 6156 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6157 // removed. 6158 srcThread->removeEffectChain_l(chain); 6159 6160 // transfer all effects one by one so that new effect chain is created on new thread with 6161 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6162 audio_io_handle_t dstOutput = dstThread->id(); 6163 sp<EffectChain> dstChain; 6164 uint32_t strategy = 0; // prevent compiler warning 6165 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6166 while (effect != 0) { 6167 srcThread->removeEffect_l(effect); 6168 dstThread->addEffect_l(effect); 6169 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6170 if (effect->state() == EffectModule::ACTIVE || 6171 effect->state() == EffectModule::STOPPING) { 6172 effect->start(); 6173 } 6174 // if the move request is not received from audio policy manager, the effect must be 6175 // re-registered with the new strategy and output 6176 if (dstChain == 0) { 6177 dstChain = effect->chain().promote(); 6178 if (dstChain == 0) { 6179 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6180 srcThread->addEffect_l(effect); 6181 return NO_INIT; 6182 } 6183 strategy = dstChain->strategy(); 6184 } 6185 if (reRegister) { 6186 AudioSystem::unregisterEffect(effect->id()); 6187 AudioSystem::registerEffect(&effect->desc(), 6188 dstOutput, 6189 strategy, 6190 sessionId, 6191 effect->id()); 6192 } 6193 effect = chain->getEffectFromId_l(0); 6194 } 6195 6196 return NO_ERROR; 6197} 6198 6199 6200// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6201sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6202 const sp<AudioFlinger::Client>& client, 6203 const sp<IEffectClient>& effectClient, 6204 int32_t priority, 6205 int sessionId, 6206 effect_descriptor_t *desc, 6207 int *enabled, 6208 status_t *status 6209 ) 6210{ 6211 sp<EffectModule> effect; 6212 sp<EffectHandle> handle; 6213 status_t lStatus; 6214 sp<EffectChain> chain; 6215 bool chainCreated = false; 6216 bool effectCreated = false; 6217 bool effectRegistered = false; 6218 6219 lStatus = initCheck(); 6220 if (lStatus != NO_ERROR) { 6221 ALOGW("createEffect_l() Audio driver not initialized."); 6222 goto Exit; 6223 } 6224 6225 // Do not allow effects with session ID 0 on direct output or duplicating threads 6226 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6227 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6228 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6229 desc->name, sessionId); 6230 lStatus = BAD_VALUE; 6231 goto Exit; 6232 } 6233 // Only Pre processor effects are allowed on input threads and only on input threads 6234 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6235 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6236 desc->name, desc->flags, mType); 6237 lStatus = BAD_VALUE; 6238 goto Exit; 6239 } 6240 6241 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6242 6243 { // scope for mLock 6244 Mutex::Autolock _l(mLock); 6245 6246 // check for existing effect chain with the requested audio session 6247 chain = getEffectChain_l(sessionId); 6248 if (chain == 0) { 6249 // create a new chain for this session 6250 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6251 chain = new EffectChain(this, sessionId); 6252 addEffectChain_l(chain); 6253 chain->setStrategy(getStrategyForSession_l(sessionId)); 6254 chainCreated = true; 6255 } else { 6256 effect = chain->getEffectFromDesc_l(desc); 6257 } 6258 6259 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6260 6261 if (effect == 0) { 6262 int id = mAudioFlinger->nextUniqueId(); 6263 // Check CPU and memory usage 6264 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6265 if (lStatus != NO_ERROR) { 6266 goto Exit; 6267 } 6268 effectRegistered = true; 6269 // create a new effect module if none present in the chain 6270 effect = new EffectModule(this, chain, desc, id, sessionId); 6271 lStatus = effect->status(); 6272 if (lStatus != NO_ERROR) { 6273 goto Exit; 6274 } 6275 lStatus = chain->addEffect_l(effect); 6276 if (lStatus != NO_ERROR) { 6277 goto Exit; 6278 } 6279 effectCreated = true; 6280 6281 effect->setDevice(mDevice); 6282 effect->setMode(mAudioFlinger->getMode()); 6283 } 6284 // create effect handle and connect it to effect module 6285 handle = new EffectHandle(effect, client, effectClient, priority); 6286 lStatus = effect->addHandle(handle); 6287 if (enabled != NULL) { 6288 *enabled = (int)effect->isEnabled(); 6289 } 6290 } 6291 6292Exit: 6293 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6294 Mutex::Autolock _l(mLock); 6295 if (effectCreated) { 6296 chain->removeEffect_l(effect); 6297 } 6298 if (effectRegistered) { 6299 AudioSystem::unregisterEffect(effect->id()); 6300 } 6301 if (chainCreated) { 6302 removeEffectChain_l(chain); 6303 } 6304 handle.clear(); 6305 } 6306 6307 if (status != NULL) { 6308 *status = lStatus; 6309 } 6310 return handle; 6311} 6312 6313sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6314{ 6315 sp<EffectChain> chain = getEffectChain_l(sessionId); 6316 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6317} 6318 6319// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6320// PlaybackThread::mLock held 6321status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6322{ 6323 // check for existing effect chain with the requested audio session 6324 int sessionId = effect->sessionId(); 6325 sp<EffectChain> chain = getEffectChain_l(sessionId); 6326 bool chainCreated = false; 6327 6328 if (chain == 0) { 6329 // create a new chain for this session 6330 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6331 chain = new EffectChain(this, sessionId); 6332 addEffectChain_l(chain); 6333 chain->setStrategy(getStrategyForSession_l(sessionId)); 6334 chainCreated = true; 6335 } 6336 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6337 6338 if (chain->getEffectFromId_l(effect->id()) != 0) { 6339 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6340 this, effect->desc().name, chain.get()); 6341 return BAD_VALUE; 6342 } 6343 6344 status_t status = chain->addEffect_l(effect); 6345 if (status != NO_ERROR) { 6346 if (chainCreated) { 6347 removeEffectChain_l(chain); 6348 } 6349 return status; 6350 } 6351 6352 effect->setDevice(mDevice); 6353 effect->setMode(mAudioFlinger->getMode()); 6354 return NO_ERROR; 6355} 6356 6357void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6358 6359 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6360 effect_descriptor_t desc = effect->desc(); 6361 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6362 detachAuxEffect_l(effect->id()); 6363 } 6364 6365 sp<EffectChain> chain = effect->chain().promote(); 6366 if (chain != 0) { 6367 // remove effect chain if removing last effect 6368 if (chain->removeEffect_l(effect) == 0) { 6369 removeEffectChain_l(chain); 6370 } 6371 } else { 6372 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6373 } 6374} 6375 6376void AudioFlinger::ThreadBase::lockEffectChains_l( 6377 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6378{ 6379 effectChains = mEffectChains; 6380 for (size_t i = 0; i < mEffectChains.size(); i++) { 6381 mEffectChains[i]->lock(); 6382 } 6383} 6384 6385void AudioFlinger::ThreadBase::unlockEffectChains( 6386 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6387{ 6388 for (size_t i = 0; i < effectChains.size(); i++) { 6389 effectChains[i]->unlock(); 6390 } 6391} 6392 6393sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6394{ 6395 Mutex::Autolock _l(mLock); 6396 return getEffectChain_l(sessionId); 6397} 6398 6399sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6400{ 6401 size_t size = mEffectChains.size(); 6402 for (size_t i = 0; i < size; i++) { 6403 if (mEffectChains[i]->sessionId() == sessionId) { 6404 return mEffectChains[i]; 6405 } 6406 } 6407 return 0; 6408} 6409 6410void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6411{ 6412 Mutex::Autolock _l(mLock); 6413 size_t size = mEffectChains.size(); 6414 for (size_t i = 0; i < size; i++) { 6415 mEffectChains[i]->setMode_l(mode); 6416 } 6417} 6418 6419void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6420 const wp<EffectHandle>& handle, 6421 bool unpinIfLast) { 6422 6423 Mutex::Autolock _l(mLock); 6424 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6425 // delete the effect module if removing last handle on it 6426 if (effect->removeHandle(handle) == 0) { 6427 if (!effect->isPinned() || unpinIfLast) { 6428 removeEffect_l(effect); 6429 AudioSystem::unregisterEffect(effect->id()); 6430 } 6431 } 6432} 6433 6434status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6435{ 6436 int session = chain->sessionId(); 6437 int16_t *buffer = mMixBuffer; 6438 bool ownsBuffer = false; 6439 6440 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6441 if (session > 0) { 6442 // Only one effect chain can be present in direct output thread and it uses 6443 // the mix buffer as input 6444 if (mType != DIRECT) { 6445 size_t numSamples = mFrameCount * mChannelCount; 6446 buffer = new int16_t[numSamples]; 6447 memset(buffer, 0, numSamples * sizeof(int16_t)); 6448 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6449 ownsBuffer = true; 6450 } 6451 6452 // Attach all tracks with same session ID to this chain. 6453 for (size_t i = 0; i < mTracks.size(); ++i) { 6454 sp<Track> track = mTracks[i]; 6455 if (session == track->sessionId()) { 6456 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6457 track->setMainBuffer(buffer); 6458 chain->incTrackCnt(); 6459 } 6460 } 6461 6462 // indicate all active tracks in the chain 6463 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6464 sp<Track> track = mActiveTracks[i].promote(); 6465 if (track == 0) continue; 6466 if (session == track->sessionId()) { 6467 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6468 chain->incActiveTrackCnt(); 6469 } 6470 } 6471 } 6472 6473 chain->setInBuffer(buffer, ownsBuffer); 6474 chain->setOutBuffer(mMixBuffer); 6475 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6476 // chains list in order to be processed last as it contains output stage effects 6477 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6478 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6479 // after track specific effects and before output stage 6480 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6481 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6482 // Effect chain for other sessions are inserted at beginning of effect 6483 // chains list to be processed before output mix effects. Relative order between other 6484 // sessions is not important 6485 size_t size = mEffectChains.size(); 6486 size_t i = 0; 6487 for (i = 0; i < size; i++) { 6488 if (mEffectChains[i]->sessionId() < session) break; 6489 } 6490 mEffectChains.insertAt(chain, i); 6491 checkSuspendOnAddEffectChain_l(chain); 6492 6493 return NO_ERROR; 6494} 6495 6496size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6497{ 6498 int session = chain->sessionId(); 6499 6500 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6501 6502 for (size_t i = 0; i < mEffectChains.size(); i++) { 6503 if (chain == mEffectChains[i]) { 6504 mEffectChains.removeAt(i); 6505 // detach all active tracks from the chain 6506 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6507 sp<Track> track = mActiveTracks[i].promote(); 6508 if (track == 0) continue; 6509 if (session == track->sessionId()) { 6510 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6511 chain.get(), session); 6512 chain->decActiveTrackCnt(); 6513 } 6514 } 6515 6516 // detach all tracks with same session ID from this chain 6517 for (size_t i = 0; i < mTracks.size(); ++i) { 6518 sp<Track> track = mTracks[i]; 6519 if (session == track->sessionId()) { 6520 track->setMainBuffer(mMixBuffer); 6521 chain->decTrackCnt(); 6522 } 6523 } 6524 break; 6525 } 6526 } 6527 return mEffectChains.size(); 6528} 6529 6530status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6531 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6532{ 6533 Mutex::Autolock _l(mLock); 6534 return attachAuxEffect_l(track, EffectId); 6535} 6536 6537status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6538 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6539{ 6540 status_t status = NO_ERROR; 6541 6542 if (EffectId == 0) { 6543 track->setAuxBuffer(0, NULL); 6544 } else { 6545 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6546 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6547 if (effect != 0) { 6548 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6549 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6550 } else { 6551 status = INVALID_OPERATION; 6552 } 6553 } else { 6554 status = BAD_VALUE; 6555 } 6556 } 6557 return status; 6558} 6559 6560void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6561{ 6562 for (size_t i = 0; i < mTracks.size(); ++i) { 6563 sp<Track> track = mTracks[i]; 6564 if (track->auxEffectId() == effectId) { 6565 attachAuxEffect_l(track, 0); 6566 } 6567 } 6568} 6569 6570status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6571{ 6572 // only one chain per input thread 6573 if (mEffectChains.size() != 0) { 6574 return INVALID_OPERATION; 6575 } 6576 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6577 6578 chain->setInBuffer(NULL); 6579 chain->setOutBuffer(NULL); 6580 6581 checkSuspendOnAddEffectChain_l(chain); 6582 6583 mEffectChains.add(chain); 6584 6585 return NO_ERROR; 6586} 6587 6588size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6589{ 6590 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6591 ALOGW_IF(mEffectChains.size() != 1, 6592 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6593 chain.get(), mEffectChains.size(), this); 6594 if (mEffectChains.size() == 1) { 6595 mEffectChains.removeAt(0); 6596 } 6597 return 0; 6598} 6599 6600// ---------------------------------------------------------------------------- 6601// EffectModule implementation 6602// ---------------------------------------------------------------------------- 6603 6604#undef LOG_TAG 6605#define LOG_TAG "AudioFlinger::EffectModule" 6606 6607AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6608 const wp<AudioFlinger::EffectChain>& chain, 6609 effect_descriptor_t *desc, 6610 int id, 6611 int sessionId) 6612 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6613 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6614{ 6615 ALOGV("Constructor %p", this); 6616 int lStatus; 6617 if (thread == NULL) { 6618 return; 6619 } 6620 6621 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6622 6623 // create effect engine from effect factory 6624 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6625 6626 if (mStatus != NO_ERROR) { 6627 return; 6628 } 6629 lStatus = init(); 6630 if (lStatus < 0) { 6631 mStatus = lStatus; 6632 goto Error; 6633 } 6634 6635 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6636 mPinned = true; 6637 } 6638 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6639 return; 6640Error: 6641 EffectRelease(mEffectInterface); 6642 mEffectInterface = NULL; 6643 ALOGV("Constructor Error %d", mStatus); 6644} 6645 6646AudioFlinger::EffectModule::~EffectModule() 6647{ 6648 ALOGV("Destructor %p", this); 6649 if (mEffectInterface != NULL) { 6650 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6651 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6652 sp<ThreadBase> thread = mThread.promote(); 6653 if (thread != 0) { 6654 audio_stream_t *stream = thread->stream(); 6655 if (stream != NULL) { 6656 stream->remove_audio_effect(stream, mEffectInterface); 6657 } 6658 } 6659 } 6660 // release effect engine 6661 EffectRelease(mEffectInterface); 6662 } 6663} 6664 6665status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6666{ 6667 status_t status; 6668 6669 Mutex::Autolock _l(mLock); 6670 int priority = handle->priority(); 6671 size_t size = mHandles.size(); 6672 sp<EffectHandle> h; 6673 size_t i; 6674 for (i = 0; i < size; i++) { 6675 h = mHandles[i].promote(); 6676 if (h == 0) continue; 6677 if (h->priority() <= priority) break; 6678 } 6679 // if inserted in first place, move effect control from previous owner to this handle 6680 if (i == 0) { 6681 bool enabled = false; 6682 if (h != 0) { 6683 enabled = h->enabled(); 6684 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6685 } 6686 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6687 status = NO_ERROR; 6688 } else { 6689 status = ALREADY_EXISTS; 6690 } 6691 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6692 mHandles.insertAt(handle, i); 6693 return status; 6694} 6695 6696size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6697{ 6698 Mutex::Autolock _l(mLock); 6699 size_t size = mHandles.size(); 6700 size_t i; 6701 for (i = 0; i < size; i++) { 6702 if (mHandles[i] == handle) break; 6703 } 6704 if (i == size) { 6705 return size; 6706 } 6707 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6708 6709 bool enabled = false; 6710 EffectHandle *hdl = handle.unsafe_get(); 6711 if (hdl != NULL) { 6712 ALOGV("removeHandle() unsafe_get OK"); 6713 enabled = hdl->enabled(); 6714 } 6715 mHandles.removeAt(i); 6716 size = mHandles.size(); 6717 // if removed from first place, move effect control from this handle to next in line 6718 if (i == 0 && size != 0) { 6719 sp<EffectHandle> h = mHandles[0].promote(); 6720 if (h != 0) { 6721 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6722 } 6723 } 6724 6725 // Prevent calls to process() and other functions on effect interface from now on. 6726 // The effect engine will be released by the destructor when the last strong reference on 6727 // this object is released which can happen after next process is called. 6728 if (size == 0 && !mPinned) { 6729 mState = DESTROYED; 6730 } 6731 6732 return size; 6733} 6734 6735sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6736{ 6737 Mutex::Autolock _l(mLock); 6738 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6739} 6740 6741void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6742{ 6743 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6744 // keep a strong reference on this EffectModule to avoid calling the 6745 // destructor before we exit 6746 sp<EffectModule> keep(this); 6747 { 6748 sp<ThreadBase> thread = mThread.promote(); 6749 if (thread != 0) { 6750 thread->disconnectEffect(keep, handle, unpinIfLast); 6751 } 6752 } 6753} 6754 6755void AudioFlinger::EffectModule::updateState() { 6756 Mutex::Autolock _l(mLock); 6757 6758 switch (mState) { 6759 case RESTART: 6760 reset_l(); 6761 // FALL THROUGH 6762 6763 case STARTING: 6764 // clear auxiliary effect input buffer for next accumulation 6765 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6766 memset(mConfig.inputCfg.buffer.raw, 6767 0, 6768 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6769 } 6770 start_l(); 6771 mState = ACTIVE; 6772 break; 6773 case STOPPING: 6774 stop_l(); 6775 mDisableWaitCnt = mMaxDisableWaitCnt; 6776 mState = STOPPED; 6777 break; 6778 case STOPPED: 6779 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6780 // turn off sequence. 6781 if (--mDisableWaitCnt == 0) { 6782 reset_l(); 6783 mState = IDLE; 6784 } 6785 break; 6786 default: //IDLE , ACTIVE, DESTROYED 6787 break; 6788 } 6789} 6790 6791void AudioFlinger::EffectModule::process() 6792{ 6793 Mutex::Autolock _l(mLock); 6794 6795 if (mState == DESTROYED || mEffectInterface == NULL || 6796 mConfig.inputCfg.buffer.raw == NULL || 6797 mConfig.outputCfg.buffer.raw == NULL) { 6798 return; 6799 } 6800 6801 if (isProcessEnabled()) { 6802 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6803 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6804 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6805 mConfig.inputCfg.buffer.s32, 6806 mConfig.inputCfg.buffer.frameCount/2); 6807 } 6808 6809 // do the actual processing in the effect engine 6810 int ret = (*mEffectInterface)->process(mEffectInterface, 6811 &mConfig.inputCfg.buffer, 6812 &mConfig.outputCfg.buffer); 6813 6814 // force transition to IDLE state when engine is ready 6815 if (mState == STOPPED && ret == -ENODATA) { 6816 mDisableWaitCnt = 1; 6817 } 6818 6819 // clear auxiliary effect input buffer for next accumulation 6820 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6821 memset(mConfig.inputCfg.buffer.raw, 0, 6822 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6823 } 6824 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6825 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6826 // If an insert effect is idle and input buffer is different from output buffer, 6827 // accumulate input onto output 6828 sp<EffectChain> chain = mChain.promote(); 6829 if (chain != 0 && chain->activeTrackCnt() != 0) { 6830 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6831 int16_t *in = mConfig.inputCfg.buffer.s16; 6832 int16_t *out = mConfig.outputCfg.buffer.s16; 6833 for (size_t i = 0; i < frameCnt; i++) { 6834 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6835 } 6836 } 6837 } 6838} 6839 6840void AudioFlinger::EffectModule::reset_l() 6841{ 6842 if (mEffectInterface == NULL) { 6843 return; 6844 } 6845 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6846} 6847 6848status_t AudioFlinger::EffectModule::configure() 6849{ 6850 uint32_t channels; 6851 if (mEffectInterface == NULL) { 6852 return NO_INIT; 6853 } 6854 6855 sp<ThreadBase> thread = mThread.promote(); 6856 if (thread == 0) { 6857 return DEAD_OBJECT; 6858 } 6859 6860 // TODO: handle configuration of effects replacing track process 6861 if (thread->channelCount() == 1) { 6862 channels = AUDIO_CHANNEL_OUT_MONO; 6863 } else { 6864 channels = AUDIO_CHANNEL_OUT_STEREO; 6865 } 6866 6867 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6868 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6869 } else { 6870 mConfig.inputCfg.channels = channels; 6871 } 6872 mConfig.outputCfg.channels = channels; 6873 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6874 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6875 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6876 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6877 mConfig.inputCfg.bufferProvider.cookie = NULL; 6878 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6879 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6880 mConfig.outputCfg.bufferProvider.cookie = NULL; 6881 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6882 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6883 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6884 // Insert effect: 6885 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6886 // always overwrites output buffer: input buffer == output buffer 6887 // - in other sessions: 6888 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6889 // other effect: overwrites output buffer: input buffer == output buffer 6890 // Auxiliary effect: 6891 // accumulates in output buffer: input buffer != output buffer 6892 // Therefore: accumulate <=> input buffer != output buffer 6893 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6894 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6895 } else { 6896 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6897 } 6898 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6899 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6900 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6901 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6902 6903 ALOGV("configure() %p thread %p buffer %p framecount %d", 6904 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6905 6906 status_t cmdStatus; 6907 uint32_t size = sizeof(int); 6908 status_t status = (*mEffectInterface)->command(mEffectInterface, 6909 EFFECT_CMD_SET_CONFIG, 6910 sizeof(effect_config_t), 6911 &mConfig, 6912 &size, 6913 &cmdStatus); 6914 if (status == 0) { 6915 status = cmdStatus; 6916 } 6917 6918 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6919 (1000 * mConfig.outputCfg.buffer.frameCount); 6920 6921 return status; 6922} 6923 6924status_t AudioFlinger::EffectModule::init() 6925{ 6926 Mutex::Autolock _l(mLock); 6927 if (mEffectInterface == NULL) { 6928 return NO_INIT; 6929 } 6930 status_t cmdStatus; 6931 uint32_t size = sizeof(status_t); 6932 status_t status = (*mEffectInterface)->command(mEffectInterface, 6933 EFFECT_CMD_INIT, 6934 0, 6935 NULL, 6936 &size, 6937 &cmdStatus); 6938 if (status == 0) { 6939 status = cmdStatus; 6940 } 6941 return status; 6942} 6943 6944status_t AudioFlinger::EffectModule::start() 6945{ 6946 Mutex::Autolock _l(mLock); 6947 return start_l(); 6948} 6949 6950status_t AudioFlinger::EffectModule::start_l() 6951{ 6952 if (mEffectInterface == NULL) { 6953 return NO_INIT; 6954 } 6955 status_t cmdStatus; 6956 uint32_t size = sizeof(status_t); 6957 status_t status = (*mEffectInterface)->command(mEffectInterface, 6958 EFFECT_CMD_ENABLE, 6959 0, 6960 NULL, 6961 &size, 6962 &cmdStatus); 6963 if (status == 0) { 6964 status = cmdStatus; 6965 } 6966 if (status == 0 && 6967 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6968 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6969 sp<ThreadBase> thread = mThread.promote(); 6970 if (thread != 0) { 6971 audio_stream_t *stream = thread->stream(); 6972 if (stream != NULL) { 6973 stream->add_audio_effect(stream, mEffectInterface); 6974 } 6975 } 6976 } 6977 return status; 6978} 6979 6980status_t AudioFlinger::EffectModule::stop() 6981{ 6982 Mutex::Autolock _l(mLock); 6983 return stop_l(); 6984} 6985 6986status_t AudioFlinger::EffectModule::stop_l() 6987{ 6988 if (mEffectInterface == NULL) { 6989 return NO_INIT; 6990 } 6991 status_t cmdStatus; 6992 uint32_t size = sizeof(status_t); 6993 status_t status = (*mEffectInterface)->command(mEffectInterface, 6994 EFFECT_CMD_DISABLE, 6995 0, 6996 NULL, 6997 &size, 6998 &cmdStatus); 6999 if (status == 0) { 7000 status = cmdStatus; 7001 } 7002 if (status == 0 && 7003 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7004 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7005 sp<ThreadBase> thread = mThread.promote(); 7006 if (thread != 0) { 7007 audio_stream_t *stream = thread->stream(); 7008 if (stream != NULL) { 7009 stream->remove_audio_effect(stream, mEffectInterface); 7010 } 7011 } 7012 } 7013 return status; 7014} 7015 7016status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7017 uint32_t cmdSize, 7018 void *pCmdData, 7019 uint32_t *replySize, 7020 void *pReplyData) 7021{ 7022 Mutex::Autolock _l(mLock); 7023// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7024 7025 if (mState == DESTROYED || mEffectInterface == NULL) { 7026 return NO_INIT; 7027 } 7028 status_t status = (*mEffectInterface)->command(mEffectInterface, 7029 cmdCode, 7030 cmdSize, 7031 pCmdData, 7032 replySize, 7033 pReplyData); 7034 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7035 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7036 for (size_t i = 1; i < mHandles.size(); i++) { 7037 sp<EffectHandle> h = mHandles[i].promote(); 7038 if (h != 0) { 7039 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7040 } 7041 } 7042 } 7043 return status; 7044} 7045 7046status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7047{ 7048 7049 Mutex::Autolock _l(mLock); 7050 ALOGV("setEnabled %p enabled %d", this, enabled); 7051 7052 if (enabled != isEnabled()) { 7053 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7054 if (enabled && status != NO_ERROR) { 7055 return status; 7056 } 7057 7058 switch (mState) { 7059 // going from disabled to enabled 7060 case IDLE: 7061 mState = STARTING; 7062 break; 7063 case STOPPED: 7064 mState = RESTART; 7065 break; 7066 case STOPPING: 7067 mState = ACTIVE; 7068 break; 7069 7070 // going from enabled to disabled 7071 case RESTART: 7072 mState = STOPPED; 7073 break; 7074 case STARTING: 7075 mState = IDLE; 7076 break; 7077 case ACTIVE: 7078 mState = STOPPING; 7079 break; 7080 case DESTROYED: 7081 return NO_ERROR; // simply ignore as we are being destroyed 7082 } 7083 for (size_t i = 1; i < mHandles.size(); i++) { 7084 sp<EffectHandle> h = mHandles[i].promote(); 7085 if (h != 0) { 7086 h->setEnabled(enabled); 7087 } 7088 } 7089 } 7090 return NO_ERROR; 7091} 7092 7093bool AudioFlinger::EffectModule::isEnabled() const 7094{ 7095 switch (mState) { 7096 case RESTART: 7097 case STARTING: 7098 case ACTIVE: 7099 return true; 7100 case IDLE: 7101 case STOPPING: 7102 case STOPPED: 7103 case DESTROYED: 7104 default: 7105 return false; 7106 } 7107} 7108 7109bool AudioFlinger::EffectModule::isProcessEnabled() const 7110{ 7111 switch (mState) { 7112 case RESTART: 7113 case ACTIVE: 7114 case STOPPING: 7115 case STOPPED: 7116 return true; 7117 case IDLE: 7118 case STARTING: 7119 case DESTROYED: 7120 default: 7121 return false; 7122 } 7123} 7124 7125status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7126{ 7127 Mutex::Autolock _l(mLock); 7128 status_t status = NO_ERROR; 7129 7130 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7131 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7132 if (isProcessEnabled() && 7133 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7134 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7135 status_t cmdStatus; 7136 uint32_t volume[2]; 7137 uint32_t *pVolume = NULL; 7138 uint32_t size = sizeof(volume); 7139 volume[0] = *left; 7140 volume[1] = *right; 7141 if (controller) { 7142 pVolume = volume; 7143 } 7144 status = (*mEffectInterface)->command(mEffectInterface, 7145 EFFECT_CMD_SET_VOLUME, 7146 size, 7147 volume, 7148 &size, 7149 pVolume); 7150 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7151 *left = volume[0]; 7152 *right = volume[1]; 7153 } 7154 } 7155 return status; 7156} 7157 7158status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7159{ 7160 Mutex::Autolock _l(mLock); 7161 status_t status = NO_ERROR; 7162 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7163 // audio pre processing modules on RecordThread can receive both output and 7164 // input device indication in the same call 7165 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7166 if (dev) { 7167 status_t cmdStatus; 7168 uint32_t size = sizeof(status_t); 7169 7170 status = (*mEffectInterface)->command(mEffectInterface, 7171 EFFECT_CMD_SET_DEVICE, 7172 sizeof(uint32_t), 7173 &dev, 7174 &size, 7175 &cmdStatus); 7176 if (status == NO_ERROR) { 7177 status = cmdStatus; 7178 } 7179 } 7180 dev = device & AUDIO_DEVICE_IN_ALL; 7181 if (dev) { 7182 status_t cmdStatus; 7183 uint32_t size = sizeof(status_t); 7184 7185 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7186 EFFECT_CMD_SET_INPUT_DEVICE, 7187 sizeof(uint32_t), 7188 &dev, 7189 &size, 7190 &cmdStatus); 7191 if (status2 == NO_ERROR) { 7192 status2 = cmdStatus; 7193 } 7194 if (status == NO_ERROR) { 7195 status = status2; 7196 } 7197 } 7198 } 7199 return status; 7200} 7201 7202status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7203{ 7204 Mutex::Autolock _l(mLock); 7205 status_t status = NO_ERROR; 7206 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7207 status_t cmdStatus; 7208 uint32_t size = sizeof(status_t); 7209 status = (*mEffectInterface)->command(mEffectInterface, 7210 EFFECT_CMD_SET_AUDIO_MODE, 7211 sizeof(audio_mode_t), 7212 &mode, 7213 &size, 7214 &cmdStatus); 7215 if (status == NO_ERROR) { 7216 status = cmdStatus; 7217 } 7218 } 7219 return status; 7220} 7221 7222void AudioFlinger::EffectModule::setSuspended(bool suspended) 7223{ 7224 Mutex::Autolock _l(mLock); 7225 mSuspended = suspended; 7226} 7227 7228bool AudioFlinger::EffectModule::suspended() const 7229{ 7230 Mutex::Autolock _l(mLock); 7231 return mSuspended; 7232} 7233 7234status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7235{ 7236 const size_t SIZE = 256; 7237 char buffer[SIZE]; 7238 String8 result; 7239 7240 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7241 result.append(buffer); 7242 7243 bool locked = tryLock(mLock); 7244 // failed to lock - AudioFlinger is probably deadlocked 7245 if (!locked) { 7246 result.append("\t\tCould not lock Fx mutex:\n"); 7247 } 7248 7249 result.append("\t\tSession Status State Engine:\n"); 7250 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7251 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7252 result.append(buffer); 7253 7254 result.append("\t\tDescriptor:\n"); 7255 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7256 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7257 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7258 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7259 result.append(buffer); 7260 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7261 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7262 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7263 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7264 result.append(buffer); 7265 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7266 mDescriptor.apiVersion, 7267 mDescriptor.flags); 7268 result.append(buffer); 7269 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7270 mDescriptor.name); 7271 result.append(buffer); 7272 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7273 mDescriptor.implementor); 7274 result.append(buffer); 7275 7276 result.append("\t\t- Input configuration:\n"); 7277 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7278 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7279 (uint32_t)mConfig.inputCfg.buffer.raw, 7280 mConfig.inputCfg.buffer.frameCount, 7281 mConfig.inputCfg.samplingRate, 7282 mConfig.inputCfg.channels, 7283 mConfig.inputCfg.format); 7284 result.append(buffer); 7285 7286 result.append("\t\t- Output configuration:\n"); 7287 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7288 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7289 (uint32_t)mConfig.outputCfg.buffer.raw, 7290 mConfig.outputCfg.buffer.frameCount, 7291 mConfig.outputCfg.samplingRate, 7292 mConfig.outputCfg.channels, 7293 mConfig.outputCfg.format); 7294 result.append(buffer); 7295 7296 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7297 result.append(buffer); 7298 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7299 for (size_t i = 0; i < mHandles.size(); ++i) { 7300 sp<EffectHandle> handle = mHandles[i].promote(); 7301 if (handle != 0) { 7302 handle->dump(buffer, SIZE); 7303 result.append(buffer); 7304 } 7305 } 7306 7307 result.append("\n"); 7308 7309 write(fd, result.string(), result.length()); 7310 7311 if (locked) { 7312 mLock.unlock(); 7313 } 7314 7315 return NO_ERROR; 7316} 7317 7318// ---------------------------------------------------------------------------- 7319// EffectHandle implementation 7320// ---------------------------------------------------------------------------- 7321 7322#undef LOG_TAG 7323#define LOG_TAG "AudioFlinger::EffectHandle" 7324 7325AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7326 const sp<AudioFlinger::Client>& client, 7327 const sp<IEffectClient>& effectClient, 7328 int32_t priority) 7329 : BnEffect(), 7330 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7331 mPriority(priority), mHasControl(false), mEnabled(false) 7332{ 7333 ALOGV("constructor %p", this); 7334 7335 if (client == 0) { 7336 return; 7337 } 7338 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7339 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7340 if (mCblkMemory != 0) { 7341 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7342 7343 if (mCblk != NULL) { 7344 new(mCblk) effect_param_cblk_t(); 7345 mBuffer = (uint8_t *)mCblk + bufOffset; 7346 } 7347 } else { 7348 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7349 return; 7350 } 7351} 7352 7353AudioFlinger::EffectHandle::~EffectHandle() 7354{ 7355 ALOGV("Destructor %p", this); 7356 disconnect(false); 7357 ALOGV("Destructor DONE %p", this); 7358} 7359 7360status_t AudioFlinger::EffectHandle::enable() 7361{ 7362 ALOGV("enable %p", this); 7363 if (!mHasControl) return INVALID_OPERATION; 7364 if (mEffect == 0) return DEAD_OBJECT; 7365 7366 if (mEnabled) { 7367 return NO_ERROR; 7368 } 7369 7370 mEnabled = true; 7371 7372 sp<ThreadBase> thread = mEffect->thread().promote(); 7373 if (thread != 0) { 7374 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7375 } 7376 7377 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7378 if (mEffect->suspended()) { 7379 return NO_ERROR; 7380 } 7381 7382 status_t status = mEffect->setEnabled(true); 7383 if (status != NO_ERROR) { 7384 if (thread != 0) { 7385 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7386 } 7387 mEnabled = false; 7388 } 7389 return status; 7390} 7391 7392status_t AudioFlinger::EffectHandle::disable() 7393{ 7394 ALOGV("disable %p", this); 7395 if (!mHasControl) return INVALID_OPERATION; 7396 if (mEffect == 0) return DEAD_OBJECT; 7397 7398 if (!mEnabled) { 7399 return NO_ERROR; 7400 } 7401 mEnabled = false; 7402 7403 if (mEffect->suspended()) { 7404 return NO_ERROR; 7405 } 7406 7407 status_t status = mEffect->setEnabled(false); 7408 7409 sp<ThreadBase> thread = mEffect->thread().promote(); 7410 if (thread != 0) { 7411 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7412 } 7413 7414 return status; 7415} 7416 7417void AudioFlinger::EffectHandle::disconnect() 7418{ 7419 disconnect(true); 7420} 7421 7422void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7423{ 7424 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7425 if (mEffect == 0) { 7426 return; 7427 } 7428 mEffect->disconnect(this, unpinIfLast); 7429 7430 if (mHasControl && mEnabled) { 7431 sp<ThreadBase> thread = mEffect->thread().promote(); 7432 if (thread != 0) { 7433 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7434 } 7435 } 7436 7437 // release sp on module => module destructor can be called now 7438 mEffect.clear(); 7439 if (mClient != 0) { 7440 if (mCblk != NULL) { 7441 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7442 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7443 } 7444 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7445 // Client destructor must run with AudioFlinger mutex locked 7446 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7447 mClient.clear(); 7448 } 7449} 7450 7451status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7452 uint32_t cmdSize, 7453 void *pCmdData, 7454 uint32_t *replySize, 7455 void *pReplyData) 7456{ 7457// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7458// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7459 7460 // only get parameter command is permitted for applications not controlling the effect 7461 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7462 return INVALID_OPERATION; 7463 } 7464 if (mEffect == 0) return DEAD_OBJECT; 7465 if (mClient == 0) return INVALID_OPERATION; 7466 7467 // handle commands that are not forwarded transparently to effect engine 7468 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7469 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7470 // no risk to block the whole media server process or mixer threads is we are stuck here 7471 Mutex::Autolock _l(mCblk->lock); 7472 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7473 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7474 mCblk->serverIndex = 0; 7475 mCblk->clientIndex = 0; 7476 return BAD_VALUE; 7477 } 7478 status_t status = NO_ERROR; 7479 while (mCblk->serverIndex < mCblk->clientIndex) { 7480 int reply; 7481 uint32_t rsize = sizeof(int); 7482 int *p = (int *)(mBuffer + mCblk->serverIndex); 7483 int size = *p++; 7484 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7485 ALOGW("command(): invalid parameter block size"); 7486 break; 7487 } 7488 effect_param_t *param = (effect_param_t *)p; 7489 if (param->psize == 0 || param->vsize == 0) { 7490 ALOGW("command(): null parameter or value size"); 7491 mCblk->serverIndex += size; 7492 continue; 7493 } 7494 uint32_t psize = sizeof(effect_param_t) + 7495 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7496 param->vsize; 7497 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7498 psize, 7499 p, 7500 &rsize, 7501 &reply); 7502 // stop at first error encountered 7503 if (ret != NO_ERROR) { 7504 status = ret; 7505 *(int *)pReplyData = reply; 7506 break; 7507 } else if (reply != NO_ERROR) { 7508 *(int *)pReplyData = reply; 7509 break; 7510 } 7511 mCblk->serverIndex += size; 7512 } 7513 mCblk->serverIndex = 0; 7514 mCblk->clientIndex = 0; 7515 return status; 7516 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7517 *(int *)pReplyData = NO_ERROR; 7518 return enable(); 7519 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7520 *(int *)pReplyData = NO_ERROR; 7521 return disable(); 7522 } 7523 7524 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7525} 7526 7527void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7528{ 7529 ALOGV("setControl %p control %d", this, hasControl); 7530 7531 mHasControl = hasControl; 7532 mEnabled = enabled; 7533 7534 if (signal && mEffectClient != 0) { 7535 mEffectClient->controlStatusChanged(hasControl); 7536 } 7537} 7538 7539void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7540 uint32_t cmdSize, 7541 void *pCmdData, 7542 uint32_t replySize, 7543 void *pReplyData) 7544{ 7545 if (mEffectClient != 0) { 7546 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7547 } 7548} 7549 7550 7551 7552void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7553{ 7554 if (mEffectClient != 0) { 7555 mEffectClient->enableStatusChanged(enabled); 7556 } 7557} 7558 7559status_t AudioFlinger::EffectHandle::onTransact( 7560 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7561{ 7562 return BnEffect::onTransact(code, data, reply, flags); 7563} 7564 7565 7566void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7567{ 7568 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7569 7570 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7571 (mClient == 0) ? getpid_cached : mClient->pid(), 7572 mPriority, 7573 mHasControl, 7574 !locked, 7575 mCblk ? mCblk->clientIndex : 0, 7576 mCblk ? mCblk->serverIndex : 0 7577 ); 7578 7579 if (locked) { 7580 mCblk->lock.unlock(); 7581 } 7582} 7583 7584#undef LOG_TAG 7585#define LOG_TAG "AudioFlinger::EffectChain" 7586 7587AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7588 int sessionId) 7589 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7590 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7591 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7592{ 7593 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7594 if (thread == NULL) { 7595 return; 7596 } 7597 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7598 thread->frameCount(); 7599} 7600 7601AudioFlinger::EffectChain::~EffectChain() 7602{ 7603 if (mOwnInBuffer) { 7604 delete mInBuffer; 7605 } 7606 7607} 7608 7609// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7610sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7611{ 7612 size_t size = mEffects.size(); 7613 7614 for (size_t i = 0; i < size; i++) { 7615 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7616 return mEffects[i]; 7617 } 7618 } 7619 return 0; 7620} 7621 7622// getEffectFromId_l() must be called with ThreadBase::mLock held 7623sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7624{ 7625 size_t size = mEffects.size(); 7626 7627 for (size_t i = 0; i < size; i++) { 7628 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7629 if (id == 0 || mEffects[i]->id() == id) { 7630 return mEffects[i]; 7631 } 7632 } 7633 return 0; 7634} 7635 7636// getEffectFromType_l() must be called with ThreadBase::mLock held 7637sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7638 const effect_uuid_t *type) 7639{ 7640 size_t size = mEffects.size(); 7641 7642 for (size_t i = 0; i < size; i++) { 7643 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7644 return mEffects[i]; 7645 } 7646 } 7647 return 0; 7648} 7649 7650// Must be called with EffectChain::mLock locked 7651void AudioFlinger::EffectChain::process_l() 7652{ 7653 sp<ThreadBase> thread = mThread.promote(); 7654 if (thread == 0) { 7655 ALOGW("process_l(): cannot promote mixer thread"); 7656 return; 7657 } 7658 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7659 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7660 // always process effects unless no more tracks are on the session and the effect tail 7661 // has been rendered 7662 bool doProcess = true; 7663 if (!isGlobalSession) { 7664 bool tracksOnSession = (trackCnt() != 0); 7665 7666 if (!tracksOnSession && mTailBufferCount == 0) { 7667 doProcess = false; 7668 } 7669 7670 if (activeTrackCnt() == 0) { 7671 // if no track is active and the effect tail has not been rendered, 7672 // the input buffer must be cleared here as the mixer process will not do it 7673 if (tracksOnSession || mTailBufferCount > 0) { 7674 size_t numSamples = thread->frameCount() * thread->channelCount(); 7675 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7676 if (mTailBufferCount > 0) { 7677 mTailBufferCount--; 7678 } 7679 } 7680 } 7681 } 7682 7683 size_t size = mEffects.size(); 7684 if (doProcess) { 7685 for (size_t i = 0; i < size; i++) { 7686 mEffects[i]->process(); 7687 } 7688 } 7689 for (size_t i = 0; i < size; i++) { 7690 mEffects[i]->updateState(); 7691 } 7692} 7693 7694// addEffect_l() must be called with PlaybackThread::mLock held 7695status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7696{ 7697 effect_descriptor_t desc = effect->desc(); 7698 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7699 7700 Mutex::Autolock _l(mLock); 7701 effect->setChain(this); 7702 sp<ThreadBase> thread = mThread.promote(); 7703 if (thread == 0) { 7704 return NO_INIT; 7705 } 7706 effect->setThread(thread); 7707 7708 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7709 // Auxiliary effects are inserted at the beginning of mEffects vector as 7710 // they are processed first and accumulated in chain input buffer 7711 mEffects.insertAt(effect, 0); 7712 7713 // the input buffer for auxiliary effect contains mono samples in 7714 // 32 bit format. This is to avoid saturation in AudoMixer 7715 // accumulation stage. Saturation is done in EffectModule::process() before 7716 // calling the process in effect engine 7717 size_t numSamples = thread->frameCount(); 7718 int32_t *buffer = new int32_t[numSamples]; 7719 memset(buffer, 0, numSamples * sizeof(int32_t)); 7720 effect->setInBuffer((int16_t *)buffer); 7721 // auxiliary effects output samples to chain input buffer for further processing 7722 // by insert effects 7723 effect->setOutBuffer(mInBuffer); 7724 } else { 7725 // Insert effects are inserted at the end of mEffects vector as they are processed 7726 // after track and auxiliary effects. 7727 // Insert effect order as a function of indicated preference: 7728 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7729 // another effect is present 7730 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7731 // last effect claiming first position 7732 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7733 // first effect claiming last position 7734 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7735 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7736 // already present 7737 7738 size_t size = mEffects.size(); 7739 size_t idx_insert = size; 7740 ssize_t idx_insert_first = -1; 7741 ssize_t idx_insert_last = -1; 7742 7743 for (size_t i = 0; i < size; i++) { 7744 effect_descriptor_t d = mEffects[i]->desc(); 7745 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7746 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7747 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7748 // check invalid effect chaining combinations 7749 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7750 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7751 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7752 return INVALID_OPERATION; 7753 } 7754 // remember position of first insert effect and by default 7755 // select this as insert position for new effect 7756 if (idx_insert == size) { 7757 idx_insert = i; 7758 } 7759 // remember position of last insert effect claiming 7760 // first position 7761 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7762 idx_insert_first = i; 7763 } 7764 // remember position of first insert effect claiming 7765 // last position 7766 if (iPref == EFFECT_FLAG_INSERT_LAST && 7767 idx_insert_last == -1) { 7768 idx_insert_last = i; 7769 } 7770 } 7771 } 7772 7773 // modify idx_insert from first position if needed 7774 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7775 if (idx_insert_last != -1) { 7776 idx_insert = idx_insert_last; 7777 } else { 7778 idx_insert = size; 7779 } 7780 } else { 7781 if (idx_insert_first != -1) { 7782 idx_insert = idx_insert_first + 1; 7783 } 7784 } 7785 7786 // always read samples from chain input buffer 7787 effect->setInBuffer(mInBuffer); 7788 7789 // if last effect in the chain, output samples to chain 7790 // output buffer, otherwise to chain input buffer 7791 if (idx_insert == size) { 7792 if (idx_insert != 0) { 7793 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7794 mEffects[idx_insert-1]->configure(); 7795 } 7796 effect->setOutBuffer(mOutBuffer); 7797 } else { 7798 effect->setOutBuffer(mInBuffer); 7799 } 7800 mEffects.insertAt(effect, idx_insert); 7801 7802 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7803 } 7804 effect->configure(); 7805 return NO_ERROR; 7806} 7807 7808// removeEffect_l() must be called with PlaybackThread::mLock held 7809size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7810{ 7811 Mutex::Autolock _l(mLock); 7812 size_t size = mEffects.size(); 7813 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7814 7815 for (size_t i = 0; i < size; i++) { 7816 if (effect == mEffects[i]) { 7817 // calling stop here will remove pre-processing effect from the audio HAL. 7818 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7819 // the middle of a read from audio HAL 7820 if (mEffects[i]->state() == EffectModule::ACTIVE || 7821 mEffects[i]->state() == EffectModule::STOPPING) { 7822 mEffects[i]->stop(); 7823 } 7824 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7825 delete[] effect->inBuffer(); 7826 } else { 7827 if (i == size - 1 && i != 0) { 7828 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7829 mEffects[i - 1]->configure(); 7830 } 7831 } 7832 mEffects.removeAt(i); 7833 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7834 break; 7835 } 7836 } 7837 7838 return mEffects.size(); 7839} 7840 7841// setDevice_l() must be called with PlaybackThread::mLock held 7842void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7843{ 7844 size_t size = mEffects.size(); 7845 for (size_t i = 0; i < size; i++) { 7846 mEffects[i]->setDevice(device); 7847 } 7848} 7849 7850// setMode_l() must be called with PlaybackThread::mLock held 7851void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7852{ 7853 size_t size = mEffects.size(); 7854 for (size_t i = 0; i < size; i++) { 7855 mEffects[i]->setMode(mode); 7856 } 7857} 7858 7859// setVolume_l() must be called with PlaybackThread::mLock held 7860bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7861{ 7862 uint32_t newLeft = *left; 7863 uint32_t newRight = *right; 7864 bool hasControl = false; 7865 int ctrlIdx = -1; 7866 size_t size = mEffects.size(); 7867 7868 // first update volume controller 7869 for (size_t i = size; i > 0; i--) { 7870 if (mEffects[i - 1]->isProcessEnabled() && 7871 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7872 ctrlIdx = i - 1; 7873 hasControl = true; 7874 break; 7875 } 7876 } 7877 7878 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7879 if (hasControl) { 7880 *left = mNewLeftVolume; 7881 *right = mNewRightVolume; 7882 } 7883 return hasControl; 7884 } 7885 7886 mVolumeCtrlIdx = ctrlIdx; 7887 mLeftVolume = newLeft; 7888 mRightVolume = newRight; 7889 7890 // second get volume update from volume controller 7891 if (ctrlIdx >= 0) { 7892 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7893 mNewLeftVolume = newLeft; 7894 mNewRightVolume = newRight; 7895 } 7896 // then indicate volume to all other effects in chain. 7897 // Pass altered volume to effects before volume controller 7898 // and requested volume to effects after controller 7899 uint32_t lVol = newLeft; 7900 uint32_t rVol = newRight; 7901 7902 for (size_t i = 0; i < size; i++) { 7903 if ((int)i == ctrlIdx) continue; 7904 // this also works for ctrlIdx == -1 when there is no volume controller 7905 if ((int)i > ctrlIdx) { 7906 lVol = *left; 7907 rVol = *right; 7908 } 7909 mEffects[i]->setVolume(&lVol, &rVol, false); 7910 } 7911 *left = newLeft; 7912 *right = newRight; 7913 7914 return hasControl; 7915} 7916 7917status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7918{ 7919 const size_t SIZE = 256; 7920 char buffer[SIZE]; 7921 String8 result; 7922 7923 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7924 result.append(buffer); 7925 7926 bool locked = tryLock(mLock); 7927 // failed to lock - AudioFlinger is probably deadlocked 7928 if (!locked) { 7929 result.append("\tCould not lock mutex:\n"); 7930 } 7931 7932 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7933 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7934 mEffects.size(), 7935 (uint32_t)mInBuffer, 7936 (uint32_t)mOutBuffer, 7937 mActiveTrackCnt); 7938 result.append(buffer); 7939 write(fd, result.string(), result.size()); 7940 7941 for (size_t i = 0; i < mEffects.size(); ++i) { 7942 sp<EffectModule> effect = mEffects[i]; 7943 if (effect != 0) { 7944 effect->dump(fd, args); 7945 } 7946 } 7947 7948 if (locked) { 7949 mLock.unlock(); 7950 } 7951 7952 return NO_ERROR; 7953} 7954 7955// must be called with ThreadBase::mLock held 7956void AudioFlinger::EffectChain::setEffectSuspended_l( 7957 const effect_uuid_t *type, bool suspend) 7958{ 7959 sp<SuspendedEffectDesc> desc; 7960 // use effect type UUID timelow as key as there is no real risk of identical 7961 // timeLow fields among effect type UUIDs. 7962 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7963 if (suspend) { 7964 if (index >= 0) { 7965 desc = mSuspendedEffects.valueAt(index); 7966 } else { 7967 desc = new SuspendedEffectDesc(); 7968 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7969 mSuspendedEffects.add(type->timeLow, desc); 7970 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7971 } 7972 if (desc->mRefCount++ == 0) { 7973 sp<EffectModule> effect = getEffectIfEnabled(type); 7974 if (effect != 0) { 7975 desc->mEffect = effect; 7976 effect->setSuspended(true); 7977 effect->setEnabled(false); 7978 } 7979 } 7980 } else { 7981 if (index < 0) { 7982 return; 7983 } 7984 desc = mSuspendedEffects.valueAt(index); 7985 if (desc->mRefCount <= 0) { 7986 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7987 desc->mRefCount = 1; 7988 } 7989 if (--desc->mRefCount == 0) { 7990 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7991 if (desc->mEffect != 0) { 7992 sp<EffectModule> effect = desc->mEffect.promote(); 7993 if (effect != 0) { 7994 effect->setSuspended(false); 7995 sp<EffectHandle> handle = effect->controlHandle(); 7996 if (handle != 0) { 7997 effect->setEnabled(handle->enabled()); 7998 } 7999 } 8000 desc->mEffect.clear(); 8001 } 8002 mSuspendedEffects.removeItemsAt(index); 8003 } 8004 } 8005} 8006 8007// must be called with ThreadBase::mLock held 8008void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8009{ 8010 sp<SuspendedEffectDesc> desc; 8011 8012 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8013 if (suspend) { 8014 if (index >= 0) { 8015 desc = mSuspendedEffects.valueAt(index); 8016 } else { 8017 desc = new SuspendedEffectDesc(); 8018 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8019 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8020 } 8021 if (desc->mRefCount++ == 0) { 8022 Vector< sp<EffectModule> > effects; 8023 getSuspendEligibleEffects(effects); 8024 for (size_t i = 0; i < effects.size(); i++) { 8025 setEffectSuspended_l(&effects[i]->desc().type, true); 8026 } 8027 } 8028 } else { 8029 if (index < 0) { 8030 return; 8031 } 8032 desc = mSuspendedEffects.valueAt(index); 8033 if (desc->mRefCount <= 0) { 8034 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8035 desc->mRefCount = 1; 8036 } 8037 if (--desc->mRefCount == 0) { 8038 Vector<const effect_uuid_t *> types; 8039 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8040 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8041 continue; 8042 } 8043 types.add(&mSuspendedEffects.valueAt(i)->mType); 8044 } 8045 for (size_t i = 0; i < types.size(); i++) { 8046 setEffectSuspended_l(types[i], false); 8047 } 8048 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8049 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8050 } 8051 } 8052} 8053 8054 8055// The volume effect is used for automated tests only 8056#ifndef OPENSL_ES_H_ 8057static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8058 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8059const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8060#endif //OPENSL_ES_H_ 8061 8062bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8063{ 8064 // auxiliary effects and visualizer are never suspended on output mix 8065 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8066 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8067 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8068 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8069 return false; 8070 } 8071 return true; 8072} 8073 8074void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8075{ 8076 effects.clear(); 8077 for (size_t i = 0; i < mEffects.size(); i++) { 8078 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8079 effects.add(mEffects[i]); 8080 } 8081 } 8082} 8083 8084sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8085 const effect_uuid_t *type) 8086{ 8087 sp<EffectModule> effect = getEffectFromType_l(type); 8088 return effect != 0 && effect->isEnabled() ? effect : 0; 8089} 8090 8091void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8092 bool enabled) 8093{ 8094 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8095 if (enabled) { 8096 if (index < 0) { 8097 // if the effect is not suspend check if all effects are suspended 8098 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8099 if (index < 0) { 8100 return; 8101 } 8102 if (!isEffectEligibleForSuspend(effect->desc())) { 8103 return; 8104 } 8105 setEffectSuspended_l(&effect->desc().type, enabled); 8106 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8107 if (index < 0) { 8108 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8109 return; 8110 } 8111 } 8112 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8113 effect->desc().type.timeLow); 8114 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8115 // if effect is requested to suspended but was not yet enabled, supend it now. 8116 if (desc->mEffect == 0) { 8117 desc->mEffect = effect; 8118 effect->setEnabled(false); 8119 effect->setSuspended(true); 8120 } 8121 } else { 8122 if (index < 0) { 8123 return; 8124 } 8125 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8126 effect->desc().type.timeLow); 8127 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8128 desc->mEffect.clear(); 8129 effect->setSuspended(false); 8130 } 8131} 8132 8133#undef LOG_TAG 8134#define LOG_TAG "AudioFlinger" 8135 8136// ---------------------------------------------------------------------------- 8137 8138status_t AudioFlinger::onTransact( 8139 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8140{ 8141 return BnAudioFlinger::onTransact(code, data, reply, flags); 8142} 8143 8144}; // namespace android 8145