AudioFlinger.cpp revision a011e35b22f95f558d81dc9c94b68b1465c4661d
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#undef ADD_BATTERY_DATA 41 42#ifdef ADD_BATTERY_DATA 43#include <media/IMediaPlayerService.h> 44#include <media/IMediaDeathNotifier.h> 45#endif 46 47#include <private/media/AudioTrackShared.h> 48#include <private/media/AudioEffectShared.h> 49 50#include <system/audio.h> 51#include <hardware/audio.h> 52 53#include "AudioMixer.h" 54#include "AudioFlinger.h" 55#include "ServiceUtilities.h" 56 57#include <media/EffectsFactoryApi.h> 58#include <audio_effects/effect_visualizer.h> 59#include <audio_effects/effect_ns.h> 60#include <audio_effects/effect_aec.h> 61 62#include <audio_utils/primitives.h> 63 64#include <powermanager/PowerManager.h> 65 66// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72#include <common_time/cc_helper.h> 73#include <common_time/local_clock.h> 74 75// ---------------------------------------------------------------------------- 76 77 78namespace android { 79 80static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 81static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 82 83static const float MAX_GAIN = 4096.0f; 84static const uint32_t MAX_GAIN_INT = 0x1000; 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95static const int kDumpLockRetries = 50; 96static const int kDumpLockSleepUs = 20000; 97 98// don't warn about blocked writes or record buffer overflows more often than this 99static const nsecs_t kWarningThrottleNs = seconds(5); 100 101// RecordThread loop sleep time upon application overrun or audio HAL read error 102static const int kRecordThreadSleepUs = 5000; 103 104// maximum time to wait for setParameters to complete 105static const nsecs_t kSetParametersTimeoutNs = seconds(2); 106 107// minimum sleep time for the mixer thread loop when tracks are active but in underrun 108static const uint32_t kMinThreadSleepTimeUs = 5000; 109// maximum divider applied to the active sleep time in the mixer thread loop 110static const uint32_t kMaxThreadSleepTimeShift = 2; 111 112nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 113 114// ---------------------------------------------------------------------------- 115 116#ifdef ADD_BATTERY_DATA 117// To collect the amplifier usage 118static void addBatteryData(uint32_t params) { 119 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 120 if (service == NULL) { 121 // it already logged 122 return; 123 } 124 125 service->addBatteryData(params); 126} 127#endif 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152static const char * const audio_interfaces[] = { 153 "primary", 154 "a2dp", 155 "usb", 156}; 157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 158 159// ---------------------------------------------------------------------------- 160 161AudioFlinger::AudioFlinger() 162 : BnAudioFlinger(), 163 mPrimaryHardwareDev(NULL), 164 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 165 mMasterVolume(1.0f), 166 mMasterVolumeSupportLvl(MVS_NONE), 167 mMasterMute(false), 168 mNextUniqueId(1), 169 mMode(AUDIO_MODE_INVALID), 170 mBtNrecIsOff(false) 171{ 172} 173 174void AudioFlinger::onFirstRef() 175{ 176 int rc = 0; 177 178 Mutex::Autolock _l(mLock); 179 180 /* TODO: move all this work into an Init() function */ 181 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 182 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 183 uint32_t int_val; 184 if (1 == sscanf(val_str, "%u", &int_val)) { 185 mStandbyTimeInNsecs = milliseconds(int_val); 186 ALOGI("Using %u mSec as standby time.", int_val); 187 } else { 188 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 189 ALOGI("Using default %u mSec as standby time.", 190 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 191 } 192 } 193 194 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 195 const hw_module_t *mod; 196 audio_hw_device_t *dev; 197 198 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 199 if (rc) 200 continue; 201 202 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 203 mod->name, mod->id); 204 mAudioHwDevs.push(dev); 205 206 if (mPrimaryHardwareDev == NULL) { 207 mPrimaryHardwareDev = dev; 208 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 209 mod->name, mod->id, audio_interfaces[i]); 210 } 211 } 212 213 if (mPrimaryHardwareDev == NULL) { 214 ALOGE("Primary audio interface not found"); 215 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 216 } 217 218 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 219 // primary HW dev is selected can change so these conditions might not always be equivalent. 220 // When that happens, re-visit all the code that assumes this. 221 222 AutoMutex lock(mHardwareLock); 223 224 // Determine the level of master volume support the primary audio HAL has, 225 // and set the initial master volume at the same time. 226 float initialVolume = 1.0; 227 mMasterVolumeSupportLvl = MVS_NONE; 228 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 229 audio_hw_device_t *dev = mPrimaryHardwareDev; 230 231 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 232 if ((NULL != dev->get_master_volume) && 233 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 234 mMasterVolumeSupportLvl = MVS_FULL; 235 } else { 236 mMasterVolumeSupportLvl = MVS_SETONLY; 237 initialVolume = 1.0; 238 } 239 240 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 241 if ((NULL == dev->set_master_volume) || 242 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 243 mMasterVolumeSupportLvl = MVS_NONE; 244 } 245 mHardwareStatus = AUDIO_HW_IDLE; 246 } 247 248 // Set the mode for each audio HAL, and try to set the initial volume (if 249 // supported) for all of the non-primary audio HALs. 250 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 251 audio_hw_device_t *dev = mAudioHwDevs[i]; 252 253 mHardwareStatus = AUDIO_HW_INIT; 254 rc = dev->init_check(dev); 255 mHardwareStatus = AUDIO_HW_IDLE; 256 if (rc == 0) { 257 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 258 mHardwareStatus = AUDIO_HW_SET_MODE; 259 dev->set_mode(dev, mMode); 260 261 if ((dev != mPrimaryHardwareDev) && 262 (NULL != dev->set_master_volume)) { 263 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 264 dev->set_master_volume(dev, initialVolume); 265 } 266 267 mHardwareStatus = AUDIO_HW_IDLE; 268 } 269 } 270 271 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 272 ? initialVolume 273 : 1.0; 274 mMasterVolume = initialVolume; 275 mHardwareStatus = AUDIO_HW_IDLE; 276} 277 278AudioFlinger::~AudioFlinger() 279{ 280 281 while (!mRecordThreads.isEmpty()) { 282 // closeInput() will remove first entry from mRecordThreads 283 closeInput(mRecordThreads.keyAt(0)); 284 } 285 while (!mPlaybackThreads.isEmpty()) { 286 // closeOutput() will remove first entry from mPlaybackThreads 287 closeOutput(mPlaybackThreads.keyAt(0)); 288 } 289 290 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 291 // no mHardwareLock needed, as there are no other references to this 292 audio_hw_device_close(mAudioHwDevs[i]); 293 } 294} 295 296audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 297{ 298 /* first matching HW device is returned */ 299 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 300 audio_hw_device_t *dev = mAudioHwDevs[i]; 301 if ((dev->get_supported_devices(dev) & devices) == devices) 302 return dev; 303 } 304 return NULL; 305} 306 307status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 308{ 309 const size_t SIZE = 256; 310 char buffer[SIZE]; 311 String8 result; 312 313 result.append("Clients:\n"); 314 for (size_t i = 0; i < mClients.size(); ++i) { 315 sp<Client> client = mClients.valueAt(i).promote(); 316 if (client != 0) { 317 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 318 result.append(buffer); 319 } 320 } 321 322 result.append("Global session refs:\n"); 323 result.append(" session pid count\n"); 324 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 325 AudioSessionRef *r = mAudioSessionRefs[i]; 326 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 327 result.append(buffer); 328 } 329 write(fd, result.string(), result.size()); 330 return NO_ERROR; 331} 332 333 334status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 335{ 336 const size_t SIZE = 256; 337 char buffer[SIZE]; 338 String8 result; 339 hardware_call_state hardwareStatus = mHardwareStatus; 340 341 snprintf(buffer, SIZE, "Hardware status: %d\n" 342 "Standby Time mSec: %u\n", 343 hardwareStatus, 344 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 345 result.append(buffer); 346 write(fd, result.string(), result.size()); 347 return NO_ERROR; 348} 349 350status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 351{ 352 const size_t SIZE = 256; 353 char buffer[SIZE]; 354 String8 result; 355 snprintf(buffer, SIZE, "Permission Denial: " 356 "can't dump AudioFlinger from pid=%d, uid=%d\n", 357 IPCThreadState::self()->getCallingPid(), 358 IPCThreadState::self()->getCallingUid()); 359 result.append(buffer); 360 write(fd, result.string(), result.size()); 361 return NO_ERROR; 362} 363 364static bool tryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = tryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = tryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 dumpClients(fd, args); 400 dumpInternals(fd, args); 401 402 // dump playback threads 403 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 404 mPlaybackThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump record threads 408 for (size_t i = 0; i < mRecordThreads.size(); i++) { 409 mRecordThreads.valueAt(i)->dump(fd, args); 410 } 411 412 // dump all hardware devs 413 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 414 audio_hw_device_t *dev = mAudioHwDevs[i]; 415 dev->dump(dev, fd); 416 } 417 if (locked) mLock.unlock(); 418 } 419 return NO_ERROR; 420} 421 422sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 423{ 424 // If pid is already in the mClients wp<> map, then use that entry 425 // (for which promote() is always != 0), otherwise create a new entry and Client. 426 sp<Client> client = mClients.valueFor(pid).promote(); 427 if (client == 0) { 428 client = new Client(this, pid); 429 mClients.add(pid, client); 430 } 431 432 return client; 433} 434 435// IAudioFlinger interface 436 437 438sp<IAudioTrack> AudioFlinger::createTrack( 439 pid_t pid, 440 audio_stream_type_t streamType, 441 uint32_t sampleRate, 442 audio_format_t format, 443 uint32_t channelMask, 444 int frameCount, 445 IAudioFlinger::track_flags_t flags, 446 const sp<IMemory>& sharedBuffer, 447 audio_io_handle_t output, 448 int *sessionId, 449 status_t *status) 450{ 451 sp<PlaybackThread::Track> track; 452 sp<TrackHandle> trackHandle; 453 sp<Client> client; 454 status_t lStatus; 455 int lSessionId; 456 457 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 458 // but if someone uses binder directly they could bypass that and cause us to crash 459 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 460 ALOGE("createTrack() invalid stream type %d", streamType); 461 lStatus = BAD_VALUE; 462 goto Exit; 463 } 464 465 { 466 Mutex::Autolock _l(mLock); 467 PlaybackThread *thread = checkPlaybackThread_l(output); 468 PlaybackThread *effectThread = NULL; 469 if (thread == NULL) { 470 ALOGE("unknown output thread"); 471 lStatus = BAD_VALUE; 472 goto Exit; 473 } 474 475 client = registerPid_l(pid); 476 477 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 478 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 479 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 480 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 481 if (mPlaybackThreads.keyAt(i) != output) { 482 // prevent same audio session on different output threads 483 uint32_t sessions = t->hasAudioSession(*sessionId); 484 if (sessions & PlaybackThread::TRACK_SESSION) { 485 ALOGE("createTrack() session ID %d already in use", *sessionId); 486 lStatus = BAD_VALUE; 487 goto Exit; 488 } 489 // check if an effect with same session ID is waiting for a track to be created 490 if (sessions & PlaybackThread::EFFECT_SESSION) { 491 effectThread = t.get(); 492 } 493 } 494 } 495 lSessionId = *sessionId; 496 } else { 497 // if no audio session id is provided, create one here 498 lSessionId = nextUniqueId(); 499 if (sessionId != NULL) { 500 *sessionId = lSessionId; 501 } 502 } 503 ALOGV("createTrack() lSessionId: %d", lSessionId); 504 505 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 506 track = thread->createTrack_l(client, streamType, sampleRate, format, 507 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 508 509 // move effect chain to this output thread if an effect on same session was waiting 510 // for a track to be created 511 if (lStatus == NO_ERROR && effectThread != NULL) { 512 Mutex::Autolock _dl(thread->mLock); 513 Mutex::Autolock _sl(effectThread->mLock); 514 moveEffectChain_l(lSessionId, effectThread, thread, true); 515 } 516 517 // Look for sync events awaiting for a session to be used. 518 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 519 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 520 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 521 track->setSyncEvent(mPendingSyncEvents[i]); 522 mPendingSyncEvents.removeAt(i); 523 i--; 524 } 525 } 526 } 527 } 528 if (lStatus == NO_ERROR) { 529 trackHandle = new TrackHandle(track); 530 } else { 531 // remove local strong reference to Client before deleting the Track so that the Client 532 // destructor is called by the TrackBase destructor with mLock held 533 client.clear(); 534 track.clear(); 535 } 536 537Exit: 538 if (status != NULL) { 539 *status = lStatus; 540 } 541 return trackHandle; 542} 543 544uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 545{ 546 Mutex::Autolock _l(mLock); 547 PlaybackThread *thread = checkPlaybackThread_l(output); 548 if (thread == NULL) { 549 ALOGW("sampleRate() unknown thread %d", output); 550 return 0; 551 } 552 return thread->sampleRate(); 553} 554 555int AudioFlinger::channelCount(audio_io_handle_t output) const 556{ 557 Mutex::Autolock _l(mLock); 558 PlaybackThread *thread = checkPlaybackThread_l(output); 559 if (thread == NULL) { 560 ALOGW("channelCount() unknown thread %d", output); 561 return 0; 562 } 563 return thread->channelCount(); 564} 565 566audio_format_t AudioFlinger::format(audio_io_handle_t output) const 567{ 568 Mutex::Autolock _l(mLock); 569 PlaybackThread *thread = checkPlaybackThread_l(output); 570 if (thread == NULL) { 571 ALOGW("format() unknown thread %d", output); 572 return AUDIO_FORMAT_INVALID; 573 } 574 return thread->format(); 575} 576 577size_t AudioFlinger::frameCount(audio_io_handle_t output) const 578{ 579 Mutex::Autolock _l(mLock); 580 PlaybackThread *thread = checkPlaybackThread_l(output); 581 if (thread == NULL) { 582 ALOGW("frameCount() unknown thread %d", output); 583 return 0; 584 } 585 return thread->frameCount(); 586} 587 588uint32_t AudioFlinger::latency(audio_io_handle_t output) const 589{ 590 Mutex::Autolock _l(mLock); 591 PlaybackThread *thread = checkPlaybackThread_l(output); 592 if (thread == NULL) { 593 ALOGW("latency() unknown thread %d", output); 594 return 0; 595 } 596 return thread->latency(); 597} 598 599status_t AudioFlinger::setMasterVolume(float value) 600{ 601 status_t ret = initCheck(); 602 if (ret != NO_ERROR) { 603 return ret; 604 } 605 606 // check calling permissions 607 if (!settingsAllowed()) { 608 return PERMISSION_DENIED; 609 } 610 611 float swmv = value; 612 613 // when hw supports master volume, don't scale in sw mixer 614 if (MVS_NONE != mMasterVolumeSupportLvl) { 615 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 616 AutoMutex lock(mHardwareLock); 617 audio_hw_device_t *dev = mAudioHwDevs[i]; 618 619 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 620 if (NULL != dev->set_master_volume) { 621 dev->set_master_volume(dev, value); 622 } 623 mHardwareStatus = AUDIO_HW_IDLE; 624 } 625 626 swmv = 1.0; 627 } 628 629 Mutex::Autolock _l(mLock); 630 mMasterVolume = value; 631 mMasterVolumeSW = swmv; 632 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 633 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 634 635 return NO_ERROR; 636} 637 638status_t AudioFlinger::setMode(audio_mode_t mode) 639{ 640 status_t ret = initCheck(); 641 if (ret != NO_ERROR) { 642 return ret; 643 } 644 645 // check calling permissions 646 if (!settingsAllowed()) { 647 return PERMISSION_DENIED; 648 } 649 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 650 ALOGW("Illegal value: setMode(%d)", mode); 651 return BAD_VALUE; 652 } 653 654 { // scope for the lock 655 AutoMutex lock(mHardwareLock); 656 mHardwareStatus = AUDIO_HW_SET_MODE; 657 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 658 mHardwareStatus = AUDIO_HW_IDLE; 659 } 660 661 if (NO_ERROR == ret) { 662 Mutex::Autolock _l(mLock); 663 mMode = mode; 664 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 665 mPlaybackThreads.valueAt(i)->setMode(mode); 666 } 667 668 return ret; 669} 670 671status_t AudioFlinger::setMicMute(bool state) 672{ 673 status_t ret = initCheck(); 674 if (ret != NO_ERROR) { 675 return ret; 676 } 677 678 // check calling permissions 679 if (!settingsAllowed()) { 680 return PERMISSION_DENIED; 681 } 682 683 AutoMutex lock(mHardwareLock); 684 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 685 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 686 mHardwareStatus = AUDIO_HW_IDLE; 687 return ret; 688} 689 690bool AudioFlinger::getMicMute() const 691{ 692 status_t ret = initCheck(); 693 if (ret != NO_ERROR) { 694 return false; 695 } 696 697 bool state = AUDIO_MODE_INVALID; 698 AutoMutex lock(mHardwareLock); 699 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 700 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 701 mHardwareStatus = AUDIO_HW_IDLE; 702 return state; 703} 704 705status_t AudioFlinger::setMasterMute(bool muted) 706{ 707 // check calling permissions 708 if (!settingsAllowed()) { 709 return PERMISSION_DENIED; 710 } 711 712 Mutex::Autolock _l(mLock); 713 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 714 mMasterMute = muted; 715 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 716 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 717 718 return NO_ERROR; 719} 720 721float AudioFlinger::masterVolume() const 722{ 723 Mutex::Autolock _l(mLock); 724 return masterVolume_l(); 725} 726 727float AudioFlinger::masterVolumeSW() const 728{ 729 Mutex::Autolock _l(mLock); 730 return masterVolumeSW_l(); 731} 732 733bool AudioFlinger::masterMute() const 734{ 735 Mutex::Autolock _l(mLock); 736 return masterMute_l(); 737} 738 739float AudioFlinger::masterVolume_l() const 740{ 741 if (MVS_FULL == mMasterVolumeSupportLvl) { 742 float ret_val; 743 AutoMutex lock(mHardwareLock); 744 745 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 746 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 747 (NULL != mPrimaryHardwareDev->get_master_volume), 748 "can't get master volume"); 749 750 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 751 mHardwareStatus = AUDIO_HW_IDLE; 752 return ret_val; 753 } 754 755 return mMasterVolume; 756} 757 758status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 759 audio_io_handle_t output) 760{ 761 // check calling permissions 762 if (!settingsAllowed()) { 763 return PERMISSION_DENIED; 764 } 765 766 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 767 ALOGE("setStreamVolume() invalid stream %d", stream); 768 return BAD_VALUE; 769 } 770 771 AutoMutex lock(mLock); 772 PlaybackThread *thread = NULL; 773 if (output) { 774 thread = checkPlaybackThread_l(output); 775 if (thread == NULL) { 776 return BAD_VALUE; 777 } 778 } 779 780 mStreamTypes[stream].volume = value; 781 782 if (thread == NULL) { 783 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 784 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 785 } 786 } else { 787 thread->setStreamVolume(stream, value); 788 } 789 790 return NO_ERROR; 791} 792 793status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 794{ 795 // check calling permissions 796 if (!settingsAllowed()) { 797 return PERMISSION_DENIED; 798 } 799 800 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 801 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 802 ALOGE("setStreamMute() invalid stream %d", stream); 803 return BAD_VALUE; 804 } 805 806 AutoMutex lock(mLock); 807 mStreamTypes[stream].mute = muted; 808 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 809 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 810 811 return NO_ERROR; 812} 813 814float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return 0.0f; 818 } 819 820 AutoMutex lock(mLock); 821 float volume; 822 if (output) { 823 PlaybackThread *thread = checkPlaybackThread_l(output); 824 if (thread == NULL) { 825 return 0.0f; 826 } 827 volume = thread->streamVolume(stream); 828 } else { 829 volume = streamVolume_l(stream); 830 } 831 832 return volume; 833} 834 835bool AudioFlinger::streamMute(audio_stream_type_t stream) const 836{ 837 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 838 return true; 839 } 840 841 AutoMutex lock(mLock); 842 return streamMute_l(stream); 843} 844 845status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 846{ 847 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 848 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 849 // check calling permissions 850 if (!settingsAllowed()) { 851 return PERMISSION_DENIED; 852 } 853 854 // ioHandle == 0 means the parameters are global to the audio hardware interface 855 if (ioHandle == 0) { 856 status_t final_result = NO_ERROR; 857 { 858 AutoMutex lock(mHardwareLock); 859 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 860 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 861 audio_hw_device_t *dev = mAudioHwDevs[i]; 862 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 863 final_result = result ?: final_result; 864 } 865 mHardwareStatus = AUDIO_HW_IDLE; 866 } 867 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 868 AudioParameter param = AudioParameter(keyValuePairs); 869 String8 value; 870 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 871 Mutex::Autolock _l(mLock); 872 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 873 if (mBtNrecIsOff != btNrecIsOff) { 874 for (size_t i = 0; i < mRecordThreads.size(); i++) { 875 sp<RecordThread> thread = mRecordThreads.valueAt(i); 876 RecordThread::RecordTrack *track = thread->track(); 877 if (track != NULL) { 878 audio_devices_t device = (audio_devices_t)( 879 thread->device() & AUDIO_DEVICE_IN_ALL); 880 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 881 thread->setEffectSuspended(FX_IID_AEC, 882 suspend, 883 track->sessionId()); 884 thread->setEffectSuspended(FX_IID_NS, 885 suspend, 886 track->sessionId()); 887 } 888 } 889 mBtNrecIsOff = btNrecIsOff; 890 } 891 } 892 return final_result; 893 } 894 895 // hold a strong ref on thread in case closeOutput() or closeInput() is called 896 // and the thread is exited once the lock is released 897 sp<ThreadBase> thread; 898 { 899 Mutex::Autolock _l(mLock); 900 thread = checkPlaybackThread_l(ioHandle); 901 if (thread == NULL) { 902 thread = checkRecordThread_l(ioHandle); 903 } else if (thread == primaryPlaybackThread_l()) { 904 // indicate output device change to all input threads for pre processing 905 AudioParameter param = AudioParameter(keyValuePairs); 906 int value; 907 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 908 (value != 0)) { 909 for (size_t i = 0; i < mRecordThreads.size(); i++) { 910 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 911 } 912 } 913 } 914 } 915 if (thread != 0) { 916 return thread->setParameters(keyValuePairs); 917 } 918 return BAD_VALUE; 919} 920 921String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 922{ 923// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 924// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 925 926 if (ioHandle == 0) { 927 String8 out_s8; 928 929 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 930 char *s; 931 { 932 AutoMutex lock(mHardwareLock); 933 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 934 audio_hw_device_t *dev = mAudioHwDevs[i]; 935 s = dev->get_parameters(dev, keys.string()); 936 mHardwareStatus = AUDIO_HW_IDLE; 937 } 938 out_s8 += String8(s ? s : ""); 939 free(s); 940 } 941 return out_s8; 942 } 943 944 Mutex::Autolock _l(mLock); 945 946 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 947 if (playbackThread != NULL) { 948 return playbackThread->getParameters(keys); 949 } 950 RecordThread *recordThread = checkRecordThread_l(ioHandle); 951 if (recordThread != NULL) { 952 return recordThread->getParameters(keys); 953 } 954 return String8(""); 955} 956 957size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 958{ 959 status_t ret = initCheck(); 960 if (ret != NO_ERROR) { 961 return 0; 962 } 963 964 AutoMutex lock(mHardwareLock); 965 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 966 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 967 mHardwareStatus = AUDIO_HW_IDLE; 968 return size; 969} 970 971unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 972{ 973 if (ioHandle == 0) { 974 return 0; 975 } 976 977 Mutex::Autolock _l(mLock); 978 979 RecordThread *recordThread = checkRecordThread_l(ioHandle); 980 if (recordThread != NULL) { 981 return recordThread->getInputFramesLost(); 982 } 983 return 0; 984} 985 986status_t AudioFlinger::setVoiceVolume(float value) 987{ 988 status_t ret = initCheck(); 989 if (ret != NO_ERROR) { 990 return ret; 991 } 992 993 // check calling permissions 994 if (!settingsAllowed()) { 995 return PERMISSION_DENIED; 996 } 997 998 AutoMutex lock(mHardwareLock); 999 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1000 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1001 mHardwareStatus = AUDIO_HW_IDLE; 1002 1003 return ret; 1004} 1005 1006status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1007 audio_io_handle_t output) const 1008{ 1009 status_t status; 1010 1011 Mutex::Autolock _l(mLock); 1012 1013 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1014 if (playbackThread != NULL) { 1015 return playbackThread->getRenderPosition(halFrames, dspFrames); 1016 } 1017 1018 return BAD_VALUE; 1019} 1020 1021void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1022{ 1023 1024 Mutex::Autolock _l(mLock); 1025 1026 pid_t pid = IPCThreadState::self()->getCallingPid(); 1027 if (mNotificationClients.indexOfKey(pid) < 0) { 1028 sp<NotificationClient> notificationClient = new NotificationClient(this, 1029 client, 1030 pid); 1031 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1032 1033 mNotificationClients.add(pid, notificationClient); 1034 1035 sp<IBinder> binder = client->asBinder(); 1036 binder->linkToDeath(notificationClient); 1037 1038 // the config change is always sent from playback or record threads to avoid deadlock 1039 // with AudioSystem::gLock 1040 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1041 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1042 } 1043 1044 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1045 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1046 } 1047 } 1048} 1049 1050void AudioFlinger::removeNotificationClient(pid_t pid) 1051{ 1052 Mutex::Autolock _l(mLock); 1053 1054 mNotificationClients.removeItem(pid); 1055 1056 ALOGV("%d died, releasing its sessions", pid); 1057 size_t num = mAudioSessionRefs.size(); 1058 bool removed = false; 1059 for (size_t i = 0; i< num; ) { 1060 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1061 ALOGV(" pid %d @ %d", ref->mPid, i); 1062 if (ref->mPid == pid) { 1063 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1064 mAudioSessionRefs.removeAt(i); 1065 delete ref; 1066 removed = true; 1067 num--; 1068 } else { 1069 i++; 1070 } 1071 } 1072 if (removed) { 1073 purgeStaleEffects_l(); 1074 } 1075} 1076 1077// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1078void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1079{ 1080 size_t size = mNotificationClients.size(); 1081 for (size_t i = 0; i < size; i++) { 1082 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1083 param2); 1084 } 1085} 1086 1087// removeClient_l() must be called with AudioFlinger::mLock held 1088void AudioFlinger::removeClient_l(pid_t pid) 1089{ 1090 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1091 mClients.removeItem(pid); 1092} 1093 1094 1095// ---------------------------------------------------------------------------- 1096 1097AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1098 uint32_t device, type_t type) 1099 : Thread(false), 1100 mType(type), 1101 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1102 // mChannelMask 1103 mChannelCount(0), 1104 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1105 mParamStatus(NO_ERROR), 1106 mStandby(false), mId(id), 1107 mDevice(device), 1108 mDeathRecipient(new PMDeathRecipient(this)) 1109{ 1110} 1111 1112AudioFlinger::ThreadBase::~ThreadBase() 1113{ 1114 mParamCond.broadcast(); 1115 // do not lock the mutex in destructor 1116 releaseWakeLock_l(); 1117 if (mPowerManager != 0) { 1118 sp<IBinder> binder = mPowerManager->asBinder(); 1119 binder->unlinkToDeath(mDeathRecipient); 1120 } 1121} 1122 1123void AudioFlinger::ThreadBase::exit() 1124{ 1125 ALOGV("ThreadBase::exit"); 1126 { 1127 // This lock prevents the following race in thread (uniprocessor for illustration): 1128 // if (!exitPending()) { 1129 // // context switch from here to exit() 1130 // // exit() calls requestExit(), what exitPending() observes 1131 // // exit() calls signal(), which is dropped since no waiters 1132 // // context switch back from exit() to here 1133 // mWaitWorkCV.wait(...); 1134 // // now thread is hung 1135 // } 1136 AutoMutex lock(mLock); 1137 requestExit(); 1138 mWaitWorkCV.signal(); 1139 } 1140 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1141 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1142 requestExitAndWait(); 1143} 1144 1145status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1146{ 1147 status_t status; 1148 1149 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1150 Mutex::Autolock _l(mLock); 1151 1152 mNewParameters.add(keyValuePairs); 1153 mWaitWorkCV.signal(); 1154 // wait condition with timeout in case the thread loop has exited 1155 // before the request could be processed 1156 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1157 status = mParamStatus; 1158 mWaitWorkCV.signal(); 1159 } else { 1160 status = TIMED_OUT; 1161 } 1162 return status; 1163} 1164 1165void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1166{ 1167 Mutex::Autolock _l(mLock); 1168 sendConfigEvent_l(event, param); 1169} 1170 1171// sendConfigEvent_l() must be called with ThreadBase::mLock held 1172void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1173{ 1174 ConfigEvent configEvent; 1175 configEvent.mEvent = event; 1176 configEvent.mParam = param; 1177 mConfigEvents.add(configEvent); 1178 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1179 mWaitWorkCV.signal(); 1180} 1181 1182void AudioFlinger::ThreadBase::processConfigEvents() 1183{ 1184 mLock.lock(); 1185 while (!mConfigEvents.isEmpty()) { 1186 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1187 ConfigEvent configEvent = mConfigEvents[0]; 1188 mConfigEvents.removeAt(0); 1189 // release mLock before locking AudioFlinger mLock: lock order is always 1190 // AudioFlinger then ThreadBase to avoid cross deadlock 1191 mLock.unlock(); 1192 mAudioFlinger->mLock.lock(); 1193 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1194 mAudioFlinger->mLock.unlock(); 1195 mLock.lock(); 1196 } 1197 mLock.unlock(); 1198} 1199 1200status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1201{ 1202 const size_t SIZE = 256; 1203 char buffer[SIZE]; 1204 String8 result; 1205 1206 bool locked = tryLock(mLock); 1207 if (!locked) { 1208 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1209 write(fd, buffer, strlen(buffer)); 1210 } 1211 1212 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1213 result.append(buffer); 1214 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1217 result.append(buffer); 1218 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1219 result.append(buffer); 1220 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1221 result.append(buffer); 1222 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1223 result.append(buffer); 1224 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1225 result.append(buffer); 1226 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1227 result.append(buffer); 1228 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1229 result.append(buffer); 1230 1231 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1232 result.append(buffer); 1233 result.append(" Index Command"); 1234 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1235 snprintf(buffer, SIZE, "\n %02d ", i); 1236 result.append(buffer); 1237 result.append(mNewParameters[i]); 1238 } 1239 1240 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1241 result.append(buffer); 1242 snprintf(buffer, SIZE, " Index event param\n"); 1243 result.append(buffer); 1244 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1245 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1246 result.append(buffer); 1247 } 1248 result.append("\n"); 1249 1250 write(fd, result.string(), result.size()); 1251 1252 if (locked) { 1253 mLock.unlock(); 1254 } 1255 return NO_ERROR; 1256} 1257 1258status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1259{ 1260 const size_t SIZE = 256; 1261 char buffer[SIZE]; 1262 String8 result; 1263 1264 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1265 write(fd, buffer, strlen(buffer)); 1266 1267 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1268 sp<EffectChain> chain = mEffectChains[i]; 1269 if (chain != 0) { 1270 chain->dump(fd, args); 1271 } 1272 } 1273 return NO_ERROR; 1274} 1275 1276void AudioFlinger::ThreadBase::acquireWakeLock() 1277{ 1278 Mutex::Autolock _l(mLock); 1279 acquireWakeLock_l(); 1280} 1281 1282void AudioFlinger::ThreadBase::acquireWakeLock_l() 1283{ 1284 if (mPowerManager == 0) { 1285 // use checkService() to avoid blocking if power service is not up yet 1286 sp<IBinder> binder = 1287 defaultServiceManager()->checkService(String16("power")); 1288 if (binder == 0) { 1289 ALOGW("Thread %s cannot connect to the power manager service", mName); 1290 } else { 1291 mPowerManager = interface_cast<IPowerManager>(binder); 1292 binder->linkToDeath(mDeathRecipient); 1293 } 1294 } 1295 if (mPowerManager != 0) { 1296 sp<IBinder> binder = new BBinder(); 1297 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1298 binder, 1299 String16(mName)); 1300 if (status == NO_ERROR) { 1301 mWakeLockToken = binder; 1302 } 1303 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1304 } 1305} 1306 1307void AudioFlinger::ThreadBase::releaseWakeLock() 1308{ 1309 Mutex::Autolock _l(mLock); 1310 releaseWakeLock_l(); 1311} 1312 1313void AudioFlinger::ThreadBase::releaseWakeLock_l() 1314{ 1315 if (mWakeLockToken != 0) { 1316 ALOGV("releaseWakeLock_l() %s", mName); 1317 if (mPowerManager != 0) { 1318 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1319 } 1320 mWakeLockToken.clear(); 1321 } 1322} 1323 1324void AudioFlinger::ThreadBase::clearPowerManager() 1325{ 1326 Mutex::Autolock _l(mLock); 1327 releaseWakeLock_l(); 1328 mPowerManager.clear(); 1329} 1330 1331void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1332{ 1333 sp<ThreadBase> thread = mThread.promote(); 1334 if (thread != 0) { 1335 thread->clearPowerManager(); 1336 } 1337 ALOGW("power manager service died !!!"); 1338} 1339 1340void AudioFlinger::ThreadBase::setEffectSuspended( 1341 const effect_uuid_t *type, bool suspend, int sessionId) 1342{ 1343 Mutex::Autolock _l(mLock); 1344 setEffectSuspended_l(type, suspend, sessionId); 1345} 1346 1347void AudioFlinger::ThreadBase::setEffectSuspended_l( 1348 const effect_uuid_t *type, bool suspend, int sessionId) 1349{ 1350 sp<EffectChain> chain = getEffectChain_l(sessionId); 1351 if (chain != 0) { 1352 if (type != NULL) { 1353 chain->setEffectSuspended_l(type, suspend); 1354 } else { 1355 chain->setEffectSuspendedAll_l(suspend); 1356 } 1357 } 1358 1359 updateSuspendedSessions_l(type, suspend, sessionId); 1360} 1361 1362void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1363{ 1364 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1365 if (index < 0) { 1366 return; 1367 } 1368 1369 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1370 mSuspendedSessions.editValueAt(index); 1371 1372 for (size_t i = 0; i < sessionEffects.size(); i++) { 1373 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1374 for (int j = 0; j < desc->mRefCount; j++) { 1375 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1376 chain->setEffectSuspendedAll_l(true); 1377 } else { 1378 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1379 desc->mType.timeLow); 1380 chain->setEffectSuspended_l(&desc->mType, true); 1381 } 1382 } 1383 } 1384} 1385 1386void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1387 bool suspend, 1388 int sessionId) 1389{ 1390 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1391 1392 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1393 1394 if (suspend) { 1395 if (index >= 0) { 1396 sessionEffects = mSuspendedSessions.editValueAt(index); 1397 } else { 1398 mSuspendedSessions.add(sessionId, sessionEffects); 1399 } 1400 } else { 1401 if (index < 0) { 1402 return; 1403 } 1404 sessionEffects = mSuspendedSessions.editValueAt(index); 1405 } 1406 1407 1408 int key = EffectChain::kKeyForSuspendAll; 1409 if (type != NULL) { 1410 key = type->timeLow; 1411 } 1412 index = sessionEffects.indexOfKey(key); 1413 1414 sp<SuspendedSessionDesc> desc; 1415 if (suspend) { 1416 if (index >= 0) { 1417 desc = sessionEffects.valueAt(index); 1418 } else { 1419 desc = new SuspendedSessionDesc(); 1420 if (type != NULL) { 1421 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1422 } 1423 sessionEffects.add(key, desc); 1424 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1425 } 1426 desc->mRefCount++; 1427 } else { 1428 if (index < 0) { 1429 return; 1430 } 1431 desc = sessionEffects.valueAt(index); 1432 if (--desc->mRefCount == 0) { 1433 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1434 sessionEffects.removeItemsAt(index); 1435 if (sessionEffects.isEmpty()) { 1436 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1437 sessionId); 1438 mSuspendedSessions.removeItem(sessionId); 1439 } 1440 } 1441 } 1442 if (!sessionEffects.isEmpty()) { 1443 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1444 } 1445} 1446 1447void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1448 bool enabled, 1449 int sessionId) 1450{ 1451 Mutex::Autolock _l(mLock); 1452 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1453} 1454 1455void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1456 bool enabled, 1457 int sessionId) 1458{ 1459 if (mType != RECORD) { 1460 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1461 // another session. This gives the priority to well behaved effect control panels 1462 // and applications not using global effects. 1463 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1464 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1465 } 1466 } 1467 1468 sp<EffectChain> chain = getEffectChain_l(sessionId); 1469 if (chain != 0) { 1470 chain->checkSuspendOnEffectEnabled(effect, enabled); 1471 } 1472} 1473 1474// ---------------------------------------------------------------------------- 1475 1476AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1477 AudioStreamOut* output, 1478 audio_io_handle_t id, 1479 uint32_t device, 1480 type_t type) 1481 : ThreadBase(audioFlinger, id, device, type), 1482 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1483 // Assumes constructor is called by AudioFlinger with it's mLock held, 1484 // but it would be safer to explicitly pass initial masterMute as parameter 1485 mMasterMute(audioFlinger->masterMute_l()), 1486 // mStreamTypes[] initialized in constructor body 1487 mOutput(output), 1488 // Assumes constructor is called by AudioFlinger with it's mLock held, 1489 // but it would be safer to explicitly pass initial masterVolume as parameter 1490 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1491 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1492 mMixerStatus(MIXER_IDLE), 1493 mPrevMixerStatus(MIXER_IDLE), 1494 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1495{ 1496 snprintf(mName, kNameLength, "AudioOut_%X", id); 1497 1498 readOutputParameters(); 1499 1500 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1501 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1502 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1503 stream = (audio_stream_type_t) (stream + 1)) { 1504 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1505 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1506 // initialized by stream_type_t default constructor 1507 // mStreamTypes[stream].valid = true; 1508 } 1509 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1510 // because mAudioFlinger doesn't have one to copy from 1511} 1512 1513AudioFlinger::PlaybackThread::~PlaybackThread() 1514{ 1515 delete [] mMixBuffer; 1516} 1517 1518status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1519{ 1520 dumpInternals(fd, args); 1521 dumpTracks(fd, args); 1522 dumpEffectChains(fd, args); 1523 return NO_ERROR; 1524} 1525 1526status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1527{ 1528 const size_t SIZE = 256; 1529 char buffer[SIZE]; 1530 String8 result; 1531 1532 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1533 result.append(buffer); 1534 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1535 for (size_t i = 0; i < mTracks.size(); ++i) { 1536 sp<Track> track = mTracks[i]; 1537 if (track != 0) { 1538 track->dump(buffer, SIZE); 1539 result.append(buffer); 1540 } 1541 } 1542 1543 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1544 result.append(buffer); 1545 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1546 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1547 sp<Track> track = mActiveTracks[i].promote(); 1548 if (track != 0) { 1549 track->dump(buffer, SIZE); 1550 result.append(buffer); 1551 } 1552 } 1553 write(fd, result.string(), result.size()); 1554 return NO_ERROR; 1555} 1556 1557status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1558{ 1559 const size_t SIZE = 256; 1560 char buffer[SIZE]; 1561 String8 result; 1562 1563 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1564 result.append(buffer); 1565 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1566 result.append(buffer); 1567 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1568 result.append(buffer); 1569 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1570 result.append(buffer); 1571 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1572 result.append(buffer); 1573 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1574 result.append(buffer); 1575 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1576 result.append(buffer); 1577 write(fd, result.string(), result.size()); 1578 1579 dumpBase(fd, args); 1580 1581 return NO_ERROR; 1582} 1583 1584// Thread virtuals 1585status_t AudioFlinger::PlaybackThread::readyToRun() 1586{ 1587 status_t status = initCheck(); 1588 if (status == NO_ERROR) { 1589 ALOGI("AudioFlinger's thread %p ready to run", this); 1590 } else { 1591 ALOGE("No working audio driver found."); 1592 } 1593 return status; 1594} 1595 1596void AudioFlinger::PlaybackThread::onFirstRef() 1597{ 1598 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1599} 1600 1601// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1602sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1603 const sp<AudioFlinger::Client>& client, 1604 audio_stream_type_t streamType, 1605 uint32_t sampleRate, 1606 audio_format_t format, 1607 uint32_t channelMask, 1608 int frameCount, 1609 const sp<IMemory>& sharedBuffer, 1610 int sessionId, 1611 bool isTimed, 1612 status_t *status) 1613{ 1614 sp<Track> track; 1615 status_t lStatus; 1616 1617 if (mType == DIRECT) { 1618 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1619 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1620 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1621 "for output %p with format %d", 1622 sampleRate, format, channelMask, mOutput, mFormat); 1623 lStatus = BAD_VALUE; 1624 goto Exit; 1625 } 1626 } 1627 } else { 1628 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1629 if (sampleRate > mSampleRate*2) { 1630 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1631 lStatus = BAD_VALUE; 1632 goto Exit; 1633 } 1634 } 1635 1636 lStatus = initCheck(); 1637 if (lStatus != NO_ERROR) { 1638 ALOGE("Audio driver not initialized."); 1639 goto Exit; 1640 } 1641 1642 { // scope for mLock 1643 Mutex::Autolock _l(mLock); 1644 1645 // all tracks in same audio session must share the same routing strategy otherwise 1646 // conflicts will happen when tracks are moved from one output to another by audio policy 1647 // manager 1648 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1649 for (size_t i = 0; i < mTracks.size(); ++i) { 1650 sp<Track> t = mTracks[i]; 1651 if (t != 0 && !t->isOutputTrack()) { 1652 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1653 if (sessionId == t->sessionId() && strategy != actual) { 1654 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1655 strategy, actual); 1656 lStatus = BAD_VALUE; 1657 goto Exit; 1658 } 1659 } 1660 } 1661 1662 if (!isTimed) { 1663 track = new Track(this, client, streamType, sampleRate, format, 1664 channelMask, frameCount, sharedBuffer, sessionId); 1665 } else { 1666 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1667 channelMask, frameCount, sharedBuffer, sessionId); 1668 } 1669 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1670 lStatus = NO_MEMORY; 1671 goto Exit; 1672 } 1673 mTracks.add(track); 1674 1675 sp<EffectChain> chain = getEffectChain_l(sessionId); 1676 if (chain != 0) { 1677 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1678 track->setMainBuffer(chain->inBuffer()); 1679 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1680 chain->incTrackCnt(); 1681 } 1682 1683 // invalidate track immediately if the stream type was moved to another thread since 1684 // createTrack() was called by the client process. 1685 if (!mStreamTypes[streamType].valid) { 1686 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1687 this, streamType); 1688 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1689 } 1690 } 1691 lStatus = NO_ERROR; 1692 1693Exit: 1694 if (status) { 1695 *status = lStatus; 1696 } 1697 return track; 1698} 1699 1700uint32_t AudioFlinger::PlaybackThread::latency() const 1701{ 1702 Mutex::Autolock _l(mLock); 1703 if (initCheck() == NO_ERROR) { 1704 return mOutput->stream->get_latency(mOutput->stream); 1705 } else { 1706 return 0; 1707 } 1708} 1709 1710void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1711{ 1712 Mutex::Autolock _l(mLock); 1713 mMasterVolume = value; 1714} 1715 1716void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1717{ 1718 Mutex::Autolock _l(mLock); 1719 setMasterMute_l(muted); 1720} 1721 1722void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1723{ 1724 Mutex::Autolock _l(mLock); 1725 mStreamTypes[stream].volume = value; 1726} 1727 1728void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1729{ 1730 Mutex::Autolock _l(mLock); 1731 mStreamTypes[stream].mute = muted; 1732} 1733 1734float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1735{ 1736 Mutex::Autolock _l(mLock); 1737 return mStreamTypes[stream].volume; 1738} 1739 1740// addTrack_l() must be called with ThreadBase::mLock held 1741status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1742{ 1743 status_t status = ALREADY_EXISTS; 1744 1745 // set retry count for buffer fill 1746 track->mRetryCount = kMaxTrackStartupRetries; 1747 if (mActiveTracks.indexOf(track) < 0) { 1748 // the track is newly added, make sure it fills up all its 1749 // buffers before playing. This is to ensure the client will 1750 // effectively get the latency it requested. 1751 track->mFillingUpStatus = Track::FS_FILLING; 1752 track->mResetDone = false; 1753 mActiveTracks.add(track); 1754 if (track->mainBuffer() != mMixBuffer) { 1755 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1756 if (chain != 0) { 1757 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1758 chain->incActiveTrackCnt(); 1759 } 1760 } 1761 1762 status = NO_ERROR; 1763 } 1764 1765 ALOGV("mWaitWorkCV.broadcast"); 1766 mWaitWorkCV.broadcast(); 1767 1768 return status; 1769} 1770 1771// destroyTrack_l() must be called with ThreadBase::mLock held 1772void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1773{ 1774 track->mState = TrackBase::TERMINATED; 1775 if (mActiveTracks.indexOf(track) < 0) { 1776 removeTrack_l(track); 1777 } 1778} 1779 1780void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1781{ 1782 mTracks.remove(track); 1783 deleteTrackName_l(track->name()); 1784 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1785 if (chain != 0) { 1786 chain->decTrackCnt(); 1787 } 1788} 1789 1790String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1791{ 1792 String8 out_s8 = String8(""); 1793 char *s; 1794 1795 Mutex::Autolock _l(mLock); 1796 if (initCheck() != NO_ERROR) { 1797 return out_s8; 1798 } 1799 1800 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1801 out_s8 = String8(s); 1802 free(s); 1803 return out_s8; 1804} 1805 1806// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1807void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1808 AudioSystem::OutputDescriptor desc; 1809 void *param2 = NULL; 1810 1811 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1812 1813 switch (event) { 1814 case AudioSystem::OUTPUT_OPENED: 1815 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1816 desc.channels = mChannelMask; 1817 desc.samplingRate = mSampleRate; 1818 desc.format = mFormat; 1819 desc.frameCount = mFrameCount; 1820 desc.latency = latency(); 1821 param2 = &desc; 1822 break; 1823 1824 case AudioSystem::STREAM_CONFIG_CHANGED: 1825 param2 = ¶m; 1826 case AudioSystem::OUTPUT_CLOSED: 1827 default: 1828 break; 1829 } 1830 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1831} 1832 1833void AudioFlinger::PlaybackThread::readOutputParameters() 1834{ 1835 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1836 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1837 mChannelCount = (uint16_t)popcount(mChannelMask); 1838 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1839 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1840 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1841 1842 // FIXME - Current mixer implementation only supports stereo output: Always 1843 // Allocate a stereo buffer even if HW output is mono. 1844 delete[] mMixBuffer; 1845 mMixBuffer = new int16_t[mFrameCount * 2]; 1846 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1847 1848 // force reconfiguration of effect chains and engines to take new buffer size and audio 1849 // parameters into account 1850 // Note that mLock is not held when readOutputParameters() is called from the constructor 1851 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1852 // matter. 1853 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1854 Vector< sp<EffectChain> > effectChains = mEffectChains; 1855 for (size_t i = 0; i < effectChains.size(); i ++) { 1856 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1857 } 1858} 1859 1860status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1861{ 1862 if (halFrames == NULL || dspFrames == NULL) { 1863 return BAD_VALUE; 1864 } 1865 Mutex::Autolock _l(mLock); 1866 if (initCheck() != NO_ERROR) { 1867 return INVALID_OPERATION; 1868 } 1869 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1870 1871 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1872} 1873 1874uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1875{ 1876 Mutex::Autolock _l(mLock); 1877 uint32_t result = 0; 1878 if (getEffectChain_l(sessionId) != 0) { 1879 result = EFFECT_SESSION; 1880 } 1881 1882 for (size_t i = 0; i < mTracks.size(); ++i) { 1883 sp<Track> track = mTracks[i]; 1884 if (sessionId == track->sessionId() && 1885 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1886 result |= TRACK_SESSION; 1887 break; 1888 } 1889 } 1890 1891 return result; 1892} 1893 1894uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1895{ 1896 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1897 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1898 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1899 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1900 } 1901 for (size_t i = 0; i < mTracks.size(); i++) { 1902 sp<Track> track = mTracks[i]; 1903 if (sessionId == track->sessionId() && 1904 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1905 return AudioSystem::getStrategyForStream(track->streamType()); 1906 } 1907 } 1908 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1909} 1910 1911 1912AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1913{ 1914 Mutex::Autolock _l(mLock); 1915 return mOutput; 1916} 1917 1918AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1919{ 1920 Mutex::Autolock _l(mLock); 1921 AudioStreamOut *output = mOutput; 1922 mOutput = NULL; 1923 return output; 1924} 1925 1926// this method must always be called either with ThreadBase mLock held or inside the thread loop 1927audio_stream_t* AudioFlinger::PlaybackThread::stream() 1928{ 1929 if (mOutput == NULL) { 1930 return NULL; 1931 } 1932 return &mOutput->stream->common; 1933} 1934 1935uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1936{ 1937 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1938 // decoding and transfer time. So sleeping for half of the latency would likely cause 1939 // underruns 1940 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1941 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1942 } else { 1943 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1944 } 1945} 1946 1947status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1948{ 1949 if (!isValidSyncEvent(event)) { 1950 return BAD_VALUE; 1951 } 1952 1953 Mutex::Autolock _l(mLock); 1954 1955 for (size_t i = 0; i < mTracks.size(); ++i) { 1956 sp<Track> track = mTracks[i]; 1957 if (event->triggerSession() == track->sessionId()) { 1958 track->setSyncEvent(event); 1959 return NO_ERROR; 1960 } 1961 } 1962 1963 return NAME_NOT_FOUND; 1964} 1965 1966bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 1967{ 1968 switch (event->type()) { 1969 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 1970 return true; 1971 default: 1972 break; 1973 } 1974 return false; 1975} 1976 1977// ---------------------------------------------------------------------------- 1978 1979AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1980 audio_io_handle_t id, uint32_t device, type_t type) 1981 : PlaybackThread(audioFlinger, output, id, device, type) 1982{ 1983 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1984 // FIXME - Current mixer implementation only supports stereo output 1985 if (mChannelCount == 1) { 1986 ALOGE("Invalid audio hardware channel count"); 1987 } 1988} 1989 1990AudioFlinger::MixerThread::~MixerThread() 1991{ 1992 delete mAudioMixer; 1993} 1994 1995class CpuStats { 1996public: 1997 CpuStats(); 1998 void sample(const String8 &title); 1999#ifdef DEBUG_CPU_USAGE 2000private: 2001 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2002 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2003 2004 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2005 2006 int mCpuNum; // thread's current CPU number 2007 int mCpukHz; // frequency of thread's current CPU in kHz 2008#endif 2009}; 2010 2011CpuStats::CpuStats() 2012#ifdef DEBUG_CPU_USAGE 2013 : mCpuNum(-1), mCpukHz(-1) 2014#endif 2015{ 2016} 2017 2018void CpuStats::sample(const String8 &title) { 2019#ifdef DEBUG_CPU_USAGE 2020 // get current thread's delta CPU time in wall clock ns 2021 double wcNs; 2022 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2023 2024 // record sample for wall clock statistics 2025 if (valid) { 2026 mWcStats.sample(wcNs); 2027 } 2028 2029 // get the current CPU number 2030 int cpuNum = sched_getcpu(); 2031 2032 // get the current CPU frequency in kHz 2033 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2034 2035 // check if either CPU number or frequency changed 2036 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2037 mCpuNum = cpuNum; 2038 mCpukHz = cpukHz; 2039 // ignore sample for purposes of cycles 2040 valid = false; 2041 } 2042 2043 // if no change in CPU number or frequency, then record sample for cycle statistics 2044 if (valid && mCpukHz > 0) { 2045 double cycles = wcNs * cpukHz * 0.000001; 2046 mHzStats.sample(cycles); 2047 } 2048 2049 unsigned n = mWcStats.n(); 2050 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2051 if ((n & 127) == 1) { 2052 long long elapsed = mCpuUsage.elapsed(); 2053 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2054 double perLoop = elapsed / (double) n; 2055 double perLoop100 = perLoop * 0.01; 2056 double perLoop1k = perLoop * 0.001; 2057 double mean = mWcStats.mean(); 2058 double stddev = mWcStats.stddev(); 2059 double minimum = mWcStats.minimum(); 2060 double maximum = mWcStats.maximum(); 2061 double meanCycles = mHzStats.mean(); 2062 double stddevCycles = mHzStats.stddev(); 2063 double minCycles = mHzStats.minimum(); 2064 double maxCycles = mHzStats.maximum(); 2065 mCpuUsage.resetElapsed(); 2066 mWcStats.reset(); 2067 mHzStats.reset(); 2068 ALOGD("CPU usage for %s over past %.1f secs\n" 2069 " (%u mixer loops at %.1f mean ms per loop):\n" 2070 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2071 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2072 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2073 title.string(), 2074 elapsed * .000000001, n, perLoop * .000001, 2075 mean * .001, 2076 stddev * .001, 2077 minimum * .001, 2078 maximum * .001, 2079 mean / perLoop100, 2080 stddev / perLoop100, 2081 minimum / perLoop100, 2082 maximum / perLoop100, 2083 meanCycles / perLoop1k, 2084 stddevCycles / perLoop1k, 2085 minCycles / perLoop1k, 2086 maxCycles / perLoop1k); 2087 2088 } 2089 } 2090#endif 2091}; 2092 2093void AudioFlinger::PlaybackThread::checkSilentMode_l() 2094{ 2095 if (!mMasterMute) { 2096 char value[PROPERTY_VALUE_MAX]; 2097 if (property_get("ro.audio.silent", value, "0") > 0) { 2098 char *endptr; 2099 unsigned long ul = strtoul(value, &endptr, 0); 2100 if (*endptr == '\0' && ul != 0) { 2101 ALOGD("Silence is golden"); 2102 // The setprop command will not allow a property to be changed after 2103 // the first time it is set, so we don't have to worry about un-muting. 2104 setMasterMute_l(true); 2105 } 2106 } 2107 } 2108} 2109 2110bool AudioFlinger::PlaybackThread::threadLoop() 2111{ 2112 Vector< sp<Track> > tracksToRemove; 2113 2114 standbyTime = systemTime(); 2115 2116 // MIXER 2117 nsecs_t lastWarning = 0; 2118if (mType == MIXER) { 2119 longStandbyExit = false; 2120} 2121 2122 // DUPLICATING 2123 // FIXME could this be made local to while loop? 2124 writeFrames = 0; 2125 2126 cacheParameters_l(); 2127 sleepTime = idleSleepTime; 2128 2129if (mType == MIXER) { 2130 sleepTimeShift = 0; 2131} 2132 2133 CpuStats cpuStats; 2134 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2135 2136 acquireWakeLock(); 2137 2138 while (!exitPending()) 2139 { 2140 cpuStats.sample(myName); 2141 2142 Vector< sp<EffectChain> > effectChains; 2143 2144 processConfigEvents(); 2145 2146 { // scope for mLock 2147 2148 Mutex::Autolock _l(mLock); 2149 2150 if (checkForNewParameters_l()) { 2151 cacheParameters_l(); 2152 } 2153 2154 saveOutputTracks(); 2155 2156 // put audio hardware into standby after short delay 2157 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2158 mSuspended > 0)) { 2159 if (!mStandby) { 2160 2161 threadLoop_standby(); 2162 2163 mStandby = true; 2164 mBytesWritten = 0; 2165 } 2166 2167 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2168 // we're about to wait, flush the binder command buffer 2169 IPCThreadState::self()->flushCommands(); 2170 2171 clearOutputTracks(); 2172 2173 if (exitPending()) break; 2174 2175 releaseWakeLock_l(); 2176 // wait until we have something to do... 2177 ALOGV("%s going to sleep", myName.string()); 2178 mWaitWorkCV.wait(mLock); 2179 ALOGV("%s waking up", myName.string()); 2180 acquireWakeLock_l(); 2181 2182 mPrevMixerStatus = MIXER_IDLE; 2183 2184 checkSilentMode_l(); 2185 2186 standbyTime = systemTime() + standbyDelay; 2187 sleepTime = idleSleepTime; 2188 if (mType == MIXER) { 2189 sleepTimeShift = 0; 2190 } 2191 2192 continue; 2193 } 2194 } 2195 2196 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2197 // Shift in the new status; this could be a queue if it's 2198 // useful to filter the mixer status over several cycles. 2199 mPrevMixerStatus = mMixerStatus; 2200 mMixerStatus = newMixerStatus; 2201 2202 // prevent any changes in effect chain list and in each effect chain 2203 // during mixing and effect process as the audio buffers could be deleted 2204 // or modified if an effect is created or deleted 2205 lockEffectChains_l(effectChains); 2206 } 2207 2208 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2209 threadLoop_mix(); 2210 } else { 2211 threadLoop_sleepTime(); 2212 } 2213 2214 if (mSuspended > 0) { 2215 sleepTime = suspendSleepTimeUs(); 2216 } 2217 2218 // only process effects if we're going to write 2219 if (sleepTime == 0) { 2220 for (size_t i = 0; i < effectChains.size(); i ++) { 2221 effectChains[i]->process_l(); 2222 } 2223 } 2224 2225 // enable changes in effect chain 2226 unlockEffectChains(effectChains); 2227 2228 // sleepTime == 0 means we must write to audio hardware 2229 if (sleepTime == 0) { 2230 2231 threadLoop_write(); 2232 2233if (mType == MIXER) { 2234 // write blocked detection 2235 nsecs_t now = systemTime(); 2236 nsecs_t delta = now - mLastWriteTime; 2237 if (!mStandby && delta > maxPeriod) { 2238 mNumDelayedWrites++; 2239 if ((now - lastWarning) > kWarningThrottleNs) { 2240 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2241 ns2ms(delta), mNumDelayedWrites, this); 2242 lastWarning = now; 2243 } 2244 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2245 // a different threshold. Or completely removed for what it is worth anyway... 2246 if (mStandby) { 2247 longStandbyExit = true; 2248 } 2249 } 2250} 2251 2252 mStandby = false; 2253 } else { 2254 usleep(sleepTime); 2255 } 2256 2257 // finally let go of removed track(s), without the lock held 2258 // since we can't guarantee the destructors won't acquire that 2259 // same lock. 2260 tracksToRemove.clear(); 2261 2262 // FIXME I don't understand the need for this here; 2263 // it was in the original code but maybe the 2264 // assignment in saveOutputTracks() makes this unnecessary? 2265 clearOutputTracks(); 2266 2267 // Effect chains will be actually deleted here if they were removed from 2268 // mEffectChains list during mixing or effects processing 2269 effectChains.clear(); 2270 2271 // FIXME Note that the above .clear() is no longer necessary since effectChains 2272 // is now local to this block, but will keep it for now (at least until merge done). 2273 } 2274 2275if (mType == MIXER || mType == DIRECT) { 2276 // put output stream into standby mode 2277 if (!mStandby) { 2278 mOutput->stream->common.standby(&mOutput->stream->common); 2279 } 2280} 2281if (mType == DUPLICATING) { 2282 // for DuplicatingThread, standby mode is handled by the outputTracks 2283} 2284 2285 releaseWakeLock(); 2286 2287 ALOGV("Thread %p type %d exiting", this, mType); 2288 return false; 2289} 2290 2291// shared by MIXER and DIRECT, overridden by DUPLICATING 2292void AudioFlinger::PlaybackThread::threadLoop_write() 2293{ 2294 // FIXME rewrite to reduce number of system calls 2295 mLastWriteTime = systemTime(); 2296 mInWrite = true; 2297 mBytesWritten += mixBufferSize; 2298 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2299 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2300 mNumWrites++; 2301 mInWrite = false; 2302} 2303 2304// shared by MIXER and DIRECT, overridden by DUPLICATING 2305void AudioFlinger::PlaybackThread::threadLoop_standby() 2306{ 2307 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2308 mOutput->stream->common.standby(&mOutput->stream->common); 2309} 2310 2311void AudioFlinger::MixerThread::threadLoop_mix() 2312{ 2313 // obtain the presentation timestamp of the next output buffer 2314 int64_t pts; 2315 status_t status = INVALID_OPERATION; 2316 2317 if (NULL != mOutput->stream->get_next_write_timestamp) { 2318 status = mOutput->stream->get_next_write_timestamp( 2319 mOutput->stream, &pts); 2320 } 2321 2322 if (status != NO_ERROR) { 2323 pts = AudioBufferProvider::kInvalidPTS; 2324 } 2325 2326 // mix buffers... 2327 mAudioMixer->process(pts); 2328 // increase sleep time progressively when application underrun condition clears. 2329 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2330 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2331 // such that we would underrun the audio HAL. 2332 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2333 sleepTimeShift--; 2334 } 2335 sleepTime = 0; 2336 standbyTime = systemTime() + standbyDelay; 2337 //TODO: delay standby when effects have a tail 2338} 2339 2340void AudioFlinger::MixerThread::threadLoop_sleepTime() 2341{ 2342 // If no tracks are ready, sleep once for the duration of an output 2343 // buffer size, then write 0s to the output 2344 if (sleepTime == 0) { 2345 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2346 sleepTime = activeSleepTime >> sleepTimeShift; 2347 if (sleepTime < kMinThreadSleepTimeUs) { 2348 sleepTime = kMinThreadSleepTimeUs; 2349 } 2350 // reduce sleep time in case of consecutive application underruns to avoid 2351 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2352 // duration we would end up writing less data than needed by the audio HAL if 2353 // the condition persists. 2354 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2355 sleepTimeShift++; 2356 } 2357 } else { 2358 sleepTime = idleSleepTime; 2359 } 2360 } else if (mBytesWritten != 0 || 2361 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2362 memset (mMixBuffer, 0, mixBufferSize); 2363 sleepTime = 0; 2364 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2365 } 2366 // TODO add standby time extension fct of effect tail 2367} 2368 2369// prepareTracks_l() must be called with ThreadBase::mLock held 2370AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2371 Vector< sp<Track> > *tracksToRemove) 2372{ 2373 2374 mixer_state mixerStatus = MIXER_IDLE; 2375 // find out which tracks need to be processed 2376 size_t count = mActiveTracks.size(); 2377 size_t mixedTracks = 0; 2378 size_t tracksWithEffect = 0; 2379 2380 float masterVolume = mMasterVolume; 2381 bool masterMute = mMasterMute; 2382 2383 if (masterMute) { 2384 masterVolume = 0; 2385 } 2386 // Delegate master volume control to effect in output mix effect chain if needed 2387 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2388 if (chain != 0) { 2389 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2390 chain->setVolume_l(&v, &v); 2391 masterVolume = (float)((v + (1 << 23)) >> 24); 2392 chain.clear(); 2393 } 2394 2395 for (size_t i=0 ; i<count ; i++) { 2396 sp<Track> t = mActiveTracks[i].promote(); 2397 if (t == 0) continue; 2398 2399 // this const just means the local variable doesn't change 2400 Track* const track = t.get(); 2401 audio_track_cblk_t* cblk = track->cblk(); 2402 2403 // The first time a track is added we wait 2404 // for all its buffers to be filled before processing it 2405 int name = track->name(); 2406 // make sure that we have enough frames to mix one full buffer. 2407 // enforce this condition only once to enable draining the buffer in case the client 2408 // app does not call stop() and relies on underrun to stop: 2409 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2410 // during last round 2411 uint32_t minFrames = 1; 2412 if (!track->isStopped() && !track->isPausing() && 2413 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2414 if (t->sampleRate() == (int)mSampleRate) { 2415 minFrames = mFrameCount; 2416 } else { 2417 // +1 for rounding and +1 for additional sample needed for interpolation 2418 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2419 // add frames already consumed but not yet released by the resampler 2420 // because cblk->framesReady() will include these frames 2421 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2422 // the minimum track buffer size is normally twice the number of frames necessary 2423 // to fill one buffer and the resampler should not leave more than one buffer worth 2424 // of unreleased frames after each pass, but just in case... 2425 ALOG_ASSERT(minFrames <= cblk->frameCount); 2426 } 2427 } 2428 if ((track->framesReady() >= minFrames) && track->isReady() && 2429 !track->isPaused() && !track->isTerminated()) 2430 { 2431 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2432 2433 mixedTracks++; 2434 2435 // track->mainBuffer() != mMixBuffer means there is an effect chain 2436 // connected to the track 2437 chain.clear(); 2438 if (track->mainBuffer() != mMixBuffer) { 2439 chain = getEffectChain_l(track->sessionId()); 2440 // Delegate volume control to effect in track effect chain if needed 2441 if (chain != 0) { 2442 tracksWithEffect++; 2443 } else { 2444 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2445 name, track->sessionId()); 2446 } 2447 } 2448 2449 2450 int param = AudioMixer::VOLUME; 2451 if (track->mFillingUpStatus == Track::FS_FILLED) { 2452 // no ramp for the first volume setting 2453 track->mFillingUpStatus = Track::FS_ACTIVE; 2454 if (track->mState == TrackBase::RESUMING) { 2455 track->mState = TrackBase::ACTIVE; 2456 param = AudioMixer::RAMP_VOLUME; 2457 } 2458 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2459 } else if (cblk->server != 0) { 2460 // If the track is stopped before the first frame was mixed, 2461 // do not apply ramp 2462 param = AudioMixer::RAMP_VOLUME; 2463 } 2464 2465 // compute volume for this track 2466 uint32_t vl, vr, va; 2467 if (track->isMuted() || track->isPausing() || 2468 mStreamTypes[track->streamType()].mute) { 2469 vl = vr = va = 0; 2470 if (track->isPausing()) { 2471 track->setPaused(); 2472 } 2473 } else { 2474 2475 // read original volumes with volume control 2476 float typeVolume = mStreamTypes[track->streamType()].volume; 2477 float v = masterVolume * typeVolume; 2478 uint32_t vlr = cblk->getVolumeLR(); 2479 vl = vlr & 0xFFFF; 2480 vr = vlr >> 16; 2481 // track volumes come from shared memory, so can't be trusted and must be clamped 2482 if (vl > MAX_GAIN_INT) { 2483 ALOGV("Track left volume out of range: %04X", vl); 2484 vl = MAX_GAIN_INT; 2485 } 2486 if (vr > MAX_GAIN_INT) { 2487 ALOGV("Track right volume out of range: %04X", vr); 2488 vr = MAX_GAIN_INT; 2489 } 2490 // now apply the master volume and stream type volume 2491 vl = (uint32_t)(v * vl) << 12; 2492 vr = (uint32_t)(v * vr) << 12; 2493 // assuming master volume and stream type volume each go up to 1.0, 2494 // vl and vr are now in 8.24 format 2495 2496 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2497 // send level comes from shared memory and so may be corrupt 2498 if (sendLevel > MAX_GAIN_INT) { 2499 ALOGV("Track send level out of range: %04X", sendLevel); 2500 sendLevel = MAX_GAIN_INT; 2501 } 2502 va = (uint32_t)(v * sendLevel); 2503 } 2504 // Delegate volume control to effect in track effect chain if needed 2505 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2506 // Do not ramp volume if volume is controlled by effect 2507 param = AudioMixer::VOLUME; 2508 track->mHasVolumeController = true; 2509 } else { 2510 // force no volume ramp when volume controller was just disabled or removed 2511 // from effect chain to avoid volume spike 2512 if (track->mHasVolumeController) { 2513 param = AudioMixer::VOLUME; 2514 } 2515 track->mHasVolumeController = false; 2516 } 2517 2518 // Convert volumes from 8.24 to 4.12 format 2519 // This additional clamping is needed in case chain->setVolume_l() overshot 2520 vl = (vl + (1 << 11)) >> 12; 2521 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2522 vr = (vr + (1 << 11)) >> 12; 2523 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2524 2525 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2526 2527 // XXX: these things DON'T need to be done each time 2528 mAudioMixer->setBufferProvider(name, track); 2529 mAudioMixer->enable(name); 2530 2531 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2532 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2533 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2534 mAudioMixer->setParameter( 2535 name, 2536 AudioMixer::TRACK, 2537 AudioMixer::FORMAT, (void *)track->format()); 2538 mAudioMixer->setParameter( 2539 name, 2540 AudioMixer::TRACK, 2541 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2542 mAudioMixer->setParameter( 2543 name, 2544 AudioMixer::RESAMPLE, 2545 AudioMixer::SAMPLE_RATE, 2546 (void *)(cblk->sampleRate)); 2547 mAudioMixer->setParameter( 2548 name, 2549 AudioMixer::TRACK, 2550 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2551 mAudioMixer->setParameter( 2552 name, 2553 AudioMixer::TRACK, 2554 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2555 2556 // reset retry count 2557 track->mRetryCount = kMaxTrackRetries; 2558 2559 // If one track is ready, set the mixer ready if: 2560 // - the mixer was not ready during previous round OR 2561 // - no other track is not ready 2562 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2563 mixerStatus != MIXER_TRACKS_ENABLED) { 2564 mixerStatus = MIXER_TRACKS_READY; 2565 } 2566 } else { 2567 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2568 if (track->isStopped()) { 2569 track->reset(); 2570 } 2571 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2572 // We have consumed all the buffers of this track. 2573 // Remove it from the list of active tracks. 2574 // TODO: use actual buffer filling status instead of latency when available from 2575 // audio HAL 2576 size_t audioHALFrames = 2577 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2578 size_t framesWritten = 2579 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2580 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2581 tracksToRemove->add(track); 2582 } 2583 } else { 2584 // No buffers for this track. Give it a few chances to 2585 // fill a buffer, then remove it from active list. 2586 if (--(track->mRetryCount) <= 0) { 2587 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2588 tracksToRemove->add(track); 2589 // indicate to client process that the track was disabled because of underrun 2590 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2591 // If one track is not ready, mark the mixer also not ready if: 2592 // - the mixer was ready during previous round OR 2593 // - no other track is ready 2594 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2595 mixerStatus != MIXER_TRACKS_READY) { 2596 mixerStatus = MIXER_TRACKS_ENABLED; 2597 } 2598 } 2599 mAudioMixer->disable(name); 2600 } 2601 } 2602 2603 // remove all the tracks that need to be... 2604 count = tracksToRemove->size(); 2605 if (CC_UNLIKELY(count)) { 2606 for (size_t i=0 ; i<count ; i++) { 2607 const sp<Track>& track = tracksToRemove->itemAt(i); 2608 mActiveTracks.remove(track); 2609 if (track->mainBuffer() != mMixBuffer) { 2610 chain = getEffectChain_l(track->sessionId()); 2611 if (chain != 0) { 2612 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2613 chain->decActiveTrackCnt(); 2614 } 2615 } 2616 if (track->isTerminated()) { 2617 removeTrack_l(track); 2618 } 2619 } 2620 } 2621 2622 // mix buffer must be cleared if all tracks are connected to an 2623 // effect chain as in this case the mixer will not write to 2624 // mix buffer and track effects will accumulate into it 2625 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2626 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2627 } 2628 2629 return mixerStatus; 2630} 2631 2632/* 2633The derived values that are cached: 2634 - mixBufferSize from frame count * frame size 2635 - activeSleepTime from activeSleepTimeUs() 2636 - idleSleepTime from idleSleepTimeUs() 2637 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2638 - maxPeriod from frame count and sample rate (MIXER only) 2639 2640The parameters that affect these derived values are: 2641 - frame count 2642 - frame size 2643 - sample rate 2644 - device type: A2DP or not 2645 - device latency 2646 - format: PCM or not 2647 - active sleep time 2648 - idle sleep time 2649*/ 2650 2651void AudioFlinger::PlaybackThread::cacheParameters_l() 2652{ 2653 mixBufferSize = mFrameCount * mFrameSize; 2654 activeSleepTime = activeSleepTimeUs(); 2655 idleSleepTime = idleSleepTimeUs(); 2656} 2657 2658void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2659{ 2660 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2661 this, streamType, mTracks.size()); 2662 Mutex::Autolock _l(mLock); 2663 2664 size_t size = mTracks.size(); 2665 for (size_t i = 0; i < size; i++) { 2666 sp<Track> t = mTracks[i]; 2667 if (t->streamType() == streamType) { 2668 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2669 t->mCblk->cv.signal(); 2670 } 2671 } 2672} 2673 2674void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2675{ 2676 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2677 this, streamType, valid); 2678 Mutex::Autolock _l(mLock); 2679 2680 mStreamTypes[streamType].valid = valid; 2681} 2682 2683// getTrackName_l() must be called with ThreadBase::mLock held 2684int AudioFlinger::MixerThread::getTrackName_l() 2685{ 2686 return mAudioMixer->getTrackName(); 2687} 2688 2689// deleteTrackName_l() must be called with ThreadBase::mLock held 2690void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2691{ 2692 ALOGV("remove track (%d) and delete from mixer", name); 2693 mAudioMixer->deleteTrackName(name); 2694} 2695 2696// checkForNewParameters_l() must be called with ThreadBase::mLock held 2697bool AudioFlinger::MixerThread::checkForNewParameters_l() 2698{ 2699 bool reconfig = false; 2700 2701 while (!mNewParameters.isEmpty()) { 2702 status_t status = NO_ERROR; 2703 String8 keyValuePair = mNewParameters[0]; 2704 AudioParameter param = AudioParameter(keyValuePair); 2705 int value; 2706 2707 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2708 reconfig = true; 2709 } 2710 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2711 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2712 status = BAD_VALUE; 2713 } else { 2714 reconfig = true; 2715 } 2716 } 2717 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2718 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2719 status = BAD_VALUE; 2720 } else { 2721 reconfig = true; 2722 } 2723 } 2724 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2725 // do not accept frame count changes if tracks are open as the track buffer 2726 // size depends on frame count and correct behavior would not be guaranteed 2727 // if frame count is changed after track creation 2728 if (!mTracks.isEmpty()) { 2729 status = INVALID_OPERATION; 2730 } else { 2731 reconfig = true; 2732 } 2733 } 2734 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2735#ifdef ADD_BATTERY_DATA 2736 // when changing the audio output device, call addBatteryData to notify 2737 // the change 2738 if ((int)mDevice != value) { 2739 uint32_t params = 0; 2740 // check whether speaker is on 2741 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2742 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2743 } 2744 2745 int deviceWithoutSpeaker 2746 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2747 // check if any other device (except speaker) is on 2748 if (value & deviceWithoutSpeaker ) { 2749 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2750 } 2751 2752 if (params != 0) { 2753 addBatteryData(params); 2754 } 2755 } 2756#endif 2757 2758 // forward device change to effects that have requested to be 2759 // aware of attached audio device. 2760 mDevice = (uint32_t)value; 2761 for (size_t i = 0; i < mEffectChains.size(); i++) { 2762 mEffectChains[i]->setDevice_l(mDevice); 2763 } 2764 } 2765 2766 if (status == NO_ERROR) { 2767 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2768 keyValuePair.string()); 2769 if (!mStandby && status == INVALID_OPERATION) { 2770 mOutput->stream->common.standby(&mOutput->stream->common); 2771 mStandby = true; 2772 mBytesWritten = 0; 2773 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2774 keyValuePair.string()); 2775 } 2776 if (status == NO_ERROR && reconfig) { 2777 delete mAudioMixer; 2778 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2779 mAudioMixer = NULL; 2780 readOutputParameters(); 2781 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2782 for (size_t i = 0; i < mTracks.size() ; i++) { 2783 int name = getTrackName_l(); 2784 if (name < 0) break; 2785 mTracks[i]->mName = name; 2786 // limit track sample rate to 2 x new output sample rate 2787 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2788 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2789 } 2790 } 2791 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2792 } 2793 } 2794 2795 mNewParameters.removeAt(0); 2796 2797 mParamStatus = status; 2798 mParamCond.signal(); 2799 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2800 // already timed out waiting for the status and will never signal the condition. 2801 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2802 } 2803 return reconfig; 2804} 2805 2806status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2807{ 2808 const size_t SIZE = 256; 2809 char buffer[SIZE]; 2810 String8 result; 2811 2812 PlaybackThread::dumpInternals(fd, args); 2813 2814 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2815 result.append(buffer); 2816 write(fd, result.string(), result.size()); 2817 return NO_ERROR; 2818} 2819 2820uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2821{ 2822 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2823} 2824 2825uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2826{ 2827 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2828} 2829 2830void AudioFlinger::MixerThread::cacheParameters_l() 2831{ 2832 PlaybackThread::cacheParameters_l(); 2833 2834 // FIXME: Relaxed timing because of a certain device that can't meet latency 2835 // Should be reduced to 2x after the vendor fixes the driver issue 2836 // increase threshold again due to low power audio mode. The way this warning 2837 // threshold is calculated and its usefulness should be reconsidered anyway. 2838 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2839} 2840 2841// ---------------------------------------------------------------------------- 2842AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2843 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2844 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2845 // mLeftVolFloat, mRightVolFloat 2846 // mLeftVolShort, mRightVolShort 2847{ 2848} 2849 2850AudioFlinger::DirectOutputThread::~DirectOutputThread() 2851{ 2852} 2853 2854AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2855 Vector< sp<Track> > *tracksToRemove 2856) 2857{ 2858 sp<Track> trackToRemove; 2859 2860 mixer_state mixerStatus = MIXER_IDLE; 2861 2862 // find out which tracks need to be processed 2863 if (mActiveTracks.size() != 0) { 2864 sp<Track> t = mActiveTracks[0].promote(); 2865 // The track died recently 2866 if (t == 0) return MIXER_IDLE; 2867 2868 Track* const track = t.get(); 2869 audio_track_cblk_t* cblk = track->cblk(); 2870 2871 // The first time a track is added we wait 2872 // for all its buffers to be filled before processing it 2873 if (cblk->framesReady() && track->isReady() && 2874 !track->isPaused() && !track->isTerminated()) 2875 { 2876 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2877 2878 if (track->mFillingUpStatus == Track::FS_FILLED) { 2879 track->mFillingUpStatus = Track::FS_ACTIVE; 2880 mLeftVolFloat = mRightVolFloat = 0; 2881 mLeftVolShort = mRightVolShort = 0; 2882 if (track->mState == TrackBase::RESUMING) { 2883 track->mState = TrackBase::ACTIVE; 2884 rampVolume = true; 2885 } 2886 } else if (cblk->server != 0) { 2887 // If the track is stopped before the first frame was mixed, 2888 // do not apply ramp 2889 rampVolume = true; 2890 } 2891 // compute volume for this track 2892 float left, right; 2893 if (track->isMuted() || mMasterMute || track->isPausing() || 2894 mStreamTypes[track->streamType()].mute) { 2895 left = right = 0; 2896 if (track->isPausing()) { 2897 track->setPaused(); 2898 } 2899 } else { 2900 float typeVolume = mStreamTypes[track->streamType()].volume; 2901 float v = mMasterVolume * typeVolume; 2902 uint32_t vlr = cblk->getVolumeLR(); 2903 float v_clamped = v * (vlr & 0xFFFF); 2904 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2905 left = v_clamped/MAX_GAIN; 2906 v_clamped = v * (vlr >> 16); 2907 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2908 right = v_clamped/MAX_GAIN; 2909 } 2910 2911 if (left != mLeftVolFloat || right != mRightVolFloat) { 2912 mLeftVolFloat = left; 2913 mRightVolFloat = right; 2914 2915 // If audio HAL implements volume control, 2916 // force software volume to nominal value 2917 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2918 left = 1.0f; 2919 right = 1.0f; 2920 } 2921 2922 // Convert volumes from float to 8.24 2923 uint32_t vl = (uint32_t)(left * (1 << 24)); 2924 uint32_t vr = (uint32_t)(right * (1 << 24)); 2925 2926 // Delegate volume control to effect in track effect chain if needed 2927 // only one effect chain can be present on DirectOutputThread, so if 2928 // there is one, the track is connected to it 2929 if (!mEffectChains.isEmpty()) { 2930 // Do not ramp volume if volume is controlled by effect 2931 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2932 rampVolume = false; 2933 } 2934 } 2935 2936 // Convert volumes from 8.24 to 4.12 format 2937 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2938 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2939 leftVol = (uint16_t)v_clamped; 2940 v_clamped = (vr + (1 << 11)) >> 12; 2941 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2942 rightVol = (uint16_t)v_clamped; 2943 } else { 2944 leftVol = mLeftVolShort; 2945 rightVol = mRightVolShort; 2946 rampVolume = false; 2947 } 2948 2949 // reset retry count 2950 track->mRetryCount = kMaxTrackRetriesDirect; 2951 mActiveTrack = t; 2952 mixerStatus = MIXER_TRACKS_READY; 2953 } else { 2954 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2955 if (track->isStopped()) { 2956 track->reset(); 2957 } 2958 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2959 // We have consumed all the buffers of this track. 2960 // Remove it from the list of active tracks. 2961 // TODO: implement behavior for compressed audio 2962 size_t audioHALFrames = 2963 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2964 size_t framesWritten = 2965 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2966 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2967 trackToRemove = track; 2968 } 2969 } else { 2970 // No buffers for this track. Give it a few chances to 2971 // fill a buffer, then remove it from active list. 2972 if (--(track->mRetryCount) <= 0) { 2973 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2974 trackToRemove = track; 2975 } else { 2976 mixerStatus = MIXER_TRACKS_ENABLED; 2977 } 2978 } 2979 } 2980 } 2981 2982 // FIXME merge this with similar code for removing multiple tracks 2983 // remove all the tracks that need to be... 2984 if (CC_UNLIKELY(trackToRemove != 0)) { 2985 tracksToRemove->add(trackToRemove); 2986 mActiveTracks.remove(trackToRemove); 2987 if (!mEffectChains.isEmpty()) { 2988 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2989 trackToRemove->sessionId()); 2990 mEffectChains[0]->decActiveTrackCnt(); 2991 } 2992 if (trackToRemove->isTerminated()) { 2993 removeTrack_l(trackToRemove); 2994 } 2995 } 2996 2997 return mixerStatus; 2998} 2999 3000void AudioFlinger::DirectOutputThread::threadLoop_mix() 3001{ 3002 AudioBufferProvider::Buffer buffer; 3003 size_t frameCount = mFrameCount; 3004 int8_t *curBuf = (int8_t *)mMixBuffer; 3005 // output audio to hardware 3006 while (frameCount) { 3007 buffer.frameCount = frameCount; 3008 mActiveTrack->getNextBuffer(&buffer); 3009 if (CC_UNLIKELY(buffer.raw == NULL)) { 3010 memset(curBuf, 0, frameCount * mFrameSize); 3011 break; 3012 } 3013 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3014 frameCount -= buffer.frameCount; 3015 curBuf += buffer.frameCount * mFrameSize; 3016 mActiveTrack->releaseBuffer(&buffer); 3017 } 3018 sleepTime = 0; 3019 standbyTime = systemTime() + standbyDelay; 3020 mActiveTrack.clear(); 3021 3022 // apply volume 3023 3024 // Do not apply volume on compressed audio 3025 if (!audio_is_linear_pcm(mFormat)) { 3026 return; 3027 } 3028 3029 // convert to signed 16 bit before volume calculation 3030 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3031 size_t count = mFrameCount * mChannelCount; 3032 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3033 int16_t *dst = mMixBuffer + count-1; 3034 while (count--) { 3035 *dst-- = (int16_t)(*src--^0x80) << 8; 3036 } 3037 } 3038 3039 frameCount = mFrameCount; 3040 int16_t *out = mMixBuffer; 3041 if (rampVolume) { 3042 if (mChannelCount == 1) { 3043 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3044 int32_t vlInc = d / (int32_t)frameCount; 3045 int32_t vl = ((int32_t)mLeftVolShort << 16); 3046 do { 3047 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3048 out++; 3049 vl += vlInc; 3050 } while (--frameCount); 3051 3052 } else { 3053 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3054 int32_t vlInc = d / (int32_t)frameCount; 3055 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3056 int32_t vrInc = d / (int32_t)frameCount; 3057 int32_t vl = ((int32_t)mLeftVolShort << 16); 3058 int32_t vr = ((int32_t)mRightVolShort << 16); 3059 do { 3060 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3061 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3062 out += 2; 3063 vl += vlInc; 3064 vr += vrInc; 3065 } while (--frameCount); 3066 } 3067 } else { 3068 if (mChannelCount == 1) { 3069 do { 3070 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3071 out++; 3072 } while (--frameCount); 3073 } else { 3074 do { 3075 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3076 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3077 out += 2; 3078 } while (--frameCount); 3079 } 3080 } 3081 3082 // convert back to unsigned 8 bit after volume calculation 3083 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3084 size_t count = mFrameCount * mChannelCount; 3085 int16_t *src = mMixBuffer; 3086 uint8_t *dst = (uint8_t *)mMixBuffer; 3087 while (count--) { 3088 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3089 } 3090 } 3091 3092 mLeftVolShort = leftVol; 3093 mRightVolShort = rightVol; 3094} 3095 3096void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3097{ 3098 if (sleepTime == 0) { 3099 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3100 sleepTime = activeSleepTime; 3101 } else { 3102 sleepTime = idleSleepTime; 3103 } 3104 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3105 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3106 sleepTime = 0; 3107 } 3108} 3109 3110// getTrackName_l() must be called with ThreadBase::mLock held 3111int AudioFlinger::DirectOutputThread::getTrackName_l() 3112{ 3113 return 0; 3114} 3115 3116// deleteTrackName_l() must be called with ThreadBase::mLock held 3117void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3118{ 3119} 3120 3121// checkForNewParameters_l() must be called with ThreadBase::mLock held 3122bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3123{ 3124 bool reconfig = false; 3125 3126 while (!mNewParameters.isEmpty()) { 3127 status_t status = NO_ERROR; 3128 String8 keyValuePair = mNewParameters[0]; 3129 AudioParameter param = AudioParameter(keyValuePair); 3130 int value; 3131 3132 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3133 // do not accept frame count changes if tracks are open as the track buffer 3134 // size depends on frame count and correct behavior would not be garantied 3135 // if frame count is changed after track creation 3136 if (!mTracks.isEmpty()) { 3137 status = INVALID_OPERATION; 3138 } else { 3139 reconfig = true; 3140 } 3141 } 3142 if (status == NO_ERROR) { 3143 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3144 keyValuePair.string()); 3145 if (!mStandby && status == INVALID_OPERATION) { 3146 mOutput->stream->common.standby(&mOutput->stream->common); 3147 mStandby = true; 3148 mBytesWritten = 0; 3149 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3150 keyValuePair.string()); 3151 } 3152 if (status == NO_ERROR && reconfig) { 3153 readOutputParameters(); 3154 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3155 } 3156 } 3157 3158 mNewParameters.removeAt(0); 3159 3160 mParamStatus = status; 3161 mParamCond.signal(); 3162 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3163 // already timed out waiting for the status and will never signal the condition. 3164 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3165 } 3166 return reconfig; 3167} 3168 3169uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3170{ 3171 uint32_t time; 3172 if (audio_is_linear_pcm(mFormat)) { 3173 time = PlaybackThread::activeSleepTimeUs(); 3174 } else { 3175 time = 10000; 3176 } 3177 return time; 3178} 3179 3180uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3181{ 3182 uint32_t time; 3183 if (audio_is_linear_pcm(mFormat)) { 3184 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3185 } else { 3186 time = 10000; 3187 } 3188 return time; 3189} 3190 3191uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3192{ 3193 uint32_t time; 3194 if (audio_is_linear_pcm(mFormat)) { 3195 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3196 } else { 3197 time = 10000; 3198 } 3199 return time; 3200} 3201 3202void AudioFlinger::DirectOutputThread::cacheParameters_l() 3203{ 3204 PlaybackThread::cacheParameters_l(); 3205 3206 // use shorter standby delay as on normal output to release 3207 // hardware resources as soon as possible 3208 standbyDelay = microseconds(activeSleepTime*2); 3209} 3210 3211// ---------------------------------------------------------------------------- 3212 3213AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3214 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3215 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3216 mWaitTimeMs(UINT_MAX) 3217{ 3218 addOutputTrack(mainThread); 3219} 3220 3221AudioFlinger::DuplicatingThread::~DuplicatingThread() 3222{ 3223 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3224 mOutputTracks[i]->destroy(); 3225 } 3226} 3227 3228void AudioFlinger::DuplicatingThread::threadLoop_mix() 3229{ 3230 // mix buffers... 3231 if (outputsReady(outputTracks)) { 3232 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3233 } else { 3234 memset(mMixBuffer, 0, mixBufferSize); 3235 } 3236 sleepTime = 0; 3237 writeFrames = mFrameCount; 3238} 3239 3240void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3241{ 3242 if (sleepTime == 0) { 3243 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3244 sleepTime = activeSleepTime; 3245 } else { 3246 sleepTime = idleSleepTime; 3247 } 3248 } else if (mBytesWritten != 0) { 3249 // flush remaining overflow buffers in output tracks 3250 for (size_t i = 0; i < outputTracks.size(); i++) { 3251 if (outputTracks[i]->isActive()) { 3252 sleepTime = 0; 3253 writeFrames = 0; 3254 memset(mMixBuffer, 0, mixBufferSize); 3255 break; 3256 } 3257 } 3258 } 3259} 3260 3261void AudioFlinger::DuplicatingThread::threadLoop_write() 3262{ 3263 standbyTime = systemTime() + standbyDelay; 3264 for (size_t i = 0; i < outputTracks.size(); i++) { 3265 outputTracks[i]->write(mMixBuffer, writeFrames); 3266 } 3267 mBytesWritten += mixBufferSize; 3268} 3269 3270void AudioFlinger::DuplicatingThread::threadLoop_standby() 3271{ 3272 // DuplicatingThread implements standby by stopping all tracks 3273 for (size_t i = 0; i < outputTracks.size(); i++) { 3274 outputTracks[i]->stop(); 3275 } 3276} 3277 3278void AudioFlinger::DuplicatingThread::saveOutputTracks() 3279{ 3280 outputTracks = mOutputTracks; 3281} 3282 3283void AudioFlinger::DuplicatingThread::clearOutputTracks() 3284{ 3285 outputTracks.clear(); 3286} 3287 3288void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3289{ 3290 Mutex::Autolock _l(mLock); 3291 // FIXME explain this formula 3292 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3293 OutputTrack *outputTrack = new OutputTrack(thread, 3294 this, 3295 mSampleRate, 3296 mFormat, 3297 mChannelMask, 3298 frameCount); 3299 if (outputTrack->cblk() != NULL) { 3300 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3301 mOutputTracks.add(outputTrack); 3302 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3303 updateWaitTime_l(); 3304 } 3305} 3306 3307void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3308{ 3309 Mutex::Autolock _l(mLock); 3310 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3311 if (mOutputTracks[i]->thread() == thread) { 3312 mOutputTracks[i]->destroy(); 3313 mOutputTracks.removeAt(i); 3314 updateWaitTime_l(); 3315 return; 3316 } 3317 } 3318 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3319} 3320 3321// caller must hold mLock 3322void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3323{ 3324 mWaitTimeMs = UINT_MAX; 3325 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3326 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3327 if (strong != 0) { 3328 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3329 if (waitTimeMs < mWaitTimeMs) { 3330 mWaitTimeMs = waitTimeMs; 3331 } 3332 } 3333 } 3334} 3335 3336 3337bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3338{ 3339 for (size_t i = 0; i < outputTracks.size(); i++) { 3340 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3341 if (thread == 0) { 3342 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3343 return false; 3344 } 3345 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3346 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3347 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3348 return false; 3349 } 3350 } 3351 return true; 3352} 3353 3354uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3355{ 3356 return (mWaitTimeMs * 1000) / 2; 3357} 3358 3359void AudioFlinger::DuplicatingThread::cacheParameters_l() 3360{ 3361 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3362 updateWaitTime_l(); 3363 3364 MixerThread::cacheParameters_l(); 3365} 3366 3367// ---------------------------------------------------------------------------- 3368 3369// TrackBase constructor must be called with AudioFlinger::mLock held 3370AudioFlinger::ThreadBase::TrackBase::TrackBase( 3371 ThreadBase *thread, 3372 const sp<Client>& client, 3373 uint32_t sampleRate, 3374 audio_format_t format, 3375 uint32_t channelMask, 3376 int frameCount, 3377 const sp<IMemory>& sharedBuffer, 3378 int sessionId) 3379 : RefBase(), 3380 mThread(thread), 3381 mClient(client), 3382 mCblk(NULL), 3383 // mBuffer 3384 // mBufferEnd 3385 mFrameCount(0), 3386 mState(IDLE), 3387 mFormat(format), 3388 mStepServerFailed(false), 3389 mSessionId(sessionId) 3390 // mChannelCount 3391 // mChannelMask 3392{ 3393 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3394 3395 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3396 size_t size = sizeof(audio_track_cblk_t); 3397 uint8_t channelCount = popcount(channelMask); 3398 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3399 if (sharedBuffer == 0) { 3400 size += bufferSize; 3401 } 3402 3403 if (client != NULL) { 3404 mCblkMemory = client->heap()->allocate(size); 3405 if (mCblkMemory != 0) { 3406 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3407 if (mCblk != NULL) { // construct the shared structure in-place. 3408 new(mCblk) audio_track_cblk_t(); 3409 // clear all buffers 3410 mCblk->frameCount = frameCount; 3411 mCblk->sampleRate = sampleRate; 3412 mChannelCount = channelCount; 3413 mChannelMask = channelMask; 3414 if (sharedBuffer == 0) { 3415 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3416 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3417 // Force underrun condition to avoid false underrun callback until first data is 3418 // written to buffer (other flags are cleared) 3419 mCblk->flags = CBLK_UNDERRUN_ON; 3420 } else { 3421 mBuffer = sharedBuffer->pointer(); 3422 } 3423 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3424 } 3425 } else { 3426 ALOGE("not enough memory for AudioTrack size=%u", size); 3427 client->heap()->dump("AudioTrack"); 3428 return; 3429 } 3430 } else { 3431 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3432 // construct the shared structure in-place. 3433 new(mCblk) audio_track_cblk_t(); 3434 // clear all buffers 3435 mCblk->frameCount = frameCount; 3436 mCblk->sampleRate = sampleRate; 3437 mChannelCount = channelCount; 3438 mChannelMask = channelMask; 3439 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3440 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3441 // Force underrun condition to avoid false underrun callback until first data is 3442 // written to buffer (other flags are cleared) 3443 mCblk->flags = CBLK_UNDERRUN_ON; 3444 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3445 } 3446} 3447 3448AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3449{ 3450 if (mCblk != NULL) { 3451 if (mClient == 0) { 3452 delete mCblk; 3453 } else { 3454 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3455 } 3456 } 3457 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3458 if (mClient != 0) { 3459 // Client destructor must run with AudioFlinger mutex locked 3460 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3461 // If the client's reference count drops to zero, the associated destructor 3462 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3463 // relying on the automatic clear() at end of scope. 3464 mClient.clear(); 3465 } 3466} 3467 3468// AudioBufferProvider interface 3469// getNextBuffer() = 0; 3470// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3471void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3472{ 3473 buffer->raw = NULL; 3474 mFrameCount = buffer->frameCount; 3475 (void) step(); // ignore return value of step() 3476 buffer->frameCount = 0; 3477} 3478 3479bool AudioFlinger::ThreadBase::TrackBase::step() { 3480 bool result; 3481 audio_track_cblk_t* cblk = this->cblk(); 3482 3483 result = cblk->stepServer(mFrameCount); 3484 if (!result) { 3485 ALOGV("stepServer failed acquiring cblk mutex"); 3486 mStepServerFailed = true; 3487 } 3488 return result; 3489} 3490 3491void AudioFlinger::ThreadBase::TrackBase::reset() { 3492 audio_track_cblk_t* cblk = this->cblk(); 3493 3494 cblk->user = 0; 3495 cblk->server = 0; 3496 cblk->userBase = 0; 3497 cblk->serverBase = 0; 3498 mStepServerFailed = false; 3499 ALOGV("TrackBase::reset"); 3500} 3501 3502int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3503 return (int)mCblk->sampleRate; 3504} 3505 3506void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3507 audio_track_cblk_t* cblk = this->cblk(); 3508 size_t frameSize = cblk->frameSize; 3509 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3510 int8_t *bufferEnd = bufferStart + frames * frameSize; 3511 3512 // Check validity of returned pointer in case the track control block would have been corrupted. 3513 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3514 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3515 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3516 server %d, serverBase %d, user %d, userBase %d", 3517 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3518 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3519 return NULL; 3520 } 3521 3522 return bufferStart; 3523} 3524 3525status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3526{ 3527 mSyncEvents.add(event); 3528 return NO_ERROR; 3529} 3530 3531// ---------------------------------------------------------------------------- 3532 3533// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3534AudioFlinger::PlaybackThread::Track::Track( 3535 PlaybackThread *thread, 3536 const sp<Client>& client, 3537 audio_stream_type_t streamType, 3538 uint32_t sampleRate, 3539 audio_format_t format, 3540 uint32_t channelMask, 3541 int frameCount, 3542 const sp<IMemory>& sharedBuffer, 3543 int sessionId) 3544 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3545 mMute(false), 3546 // mFillingUpStatus ? 3547 // mRetryCount initialized later when needed 3548 mSharedBuffer(sharedBuffer), 3549 mStreamType(streamType), 3550 mName(-1), // see note below 3551 mMainBuffer(thread->mixBuffer()), 3552 mAuxBuffer(NULL), 3553 mAuxEffectId(0), mHasVolumeController(false), 3554 mPresentationCompleteFrames(0) 3555{ 3556 if (mCblk != NULL) { 3557 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3558 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3559 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3560 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3561 mName = thread->getTrackName_l(); 3562 if (mName < 0) { 3563 ALOGE("no more track names available"); 3564 } 3565 } 3566 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3567} 3568 3569AudioFlinger::PlaybackThread::Track::~Track() 3570{ 3571 ALOGV("PlaybackThread::Track destructor"); 3572 sp<ThreadBase> thread = mThread.promote(); 3573 if (thread != 0) { 3574 Mutex::Autolock _l(thread->mLock); 3575 mState = TERMINATED; 3576 } 3577} 3578 3579void AudioFlinger::PlaybackThread::Track::destroy() 3580{ 3581 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3582 // by removing it from mTracks vector, so there is a risk that this Tracks's 3583 // destructor is called. As the destructor needs to lock mLock, 3584 // we must acquire a strong reference on this Track before locking mLock 3585 // here so that the destructor is called only when exiting this function. 3586 // On the other hand, as long as Track::destroy() is only called by 3587 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3588 // this Track with its member mTrack. 3589 sp<Track> keep(this); 3590 { // scope for mLock 3591 sp<ThreadBase> thread = mThread.promote(); 3592 if (thread != 0) { 3593 if (!isOutputTrack()) { 3594 if (mState == ACTIVE || mState == RESUMING) { 3595 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3596 3597#ifdef ADD_BATTERY_DATA 3598 // to track the speaker usage 3599 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3600#endif 3601 } 3602 AudioSystem::releaseOutput(thread->id()); 3603 } 3604 Mutex::Autolock _l(thread->mLock); 3605 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3606 playbackThread->destroyTrack_l(this); 3607 } 3608 } 3609} 3610 3611void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3612{ 3613 uint32_t vlr = mCblk->getVolumeLR(); 3614 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", 3615 mName - AudioMixer::TRACK0, 3616 (mClient == 0) ? getpid_cached : mClient->pid(), 3617 mStreamType, 3618 mFormat, 3619 mChannelMask, 3620 mSessionId, 3621 mFrameCount, 3622 mState, 3623 mMute, 3624 mFillingUpStatus, 3625 mCblk->sampleRate, 3626 vlr & 0xFFFF, 3627 vlr >> 16, 3628 mCblk->server, 3629 mCblk->user, 3630 (int)mMainBuffer, 3631 (int)mAuxBuffer); 3632} 3633 3634// AudioBufferProvider interface 3635status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3636 AudioBufferProvider::Buffer* buffer, int64_t pts) 3637{ 3638 audio_track_cblk_t* cblk = this->cblk(); 3639 uint32_t framesReady; 3640 uint32_t framesReq = buffer->frameCount; 3641 3642 // Check if last stepServer failed, try to step now 3643 if (mStepServerFailed) { 3644 if (!step()) goto getNextBuffer_exit; 3645 ALOGV("stepServer recovered"); 3646 mStepServerFailed = false; 3647 } 3648 3649 framesReady = cblk->framesReady(); 3650 3651 if (CC_LIKELY(framesReady)) { 3652 uint32_t s = cblk->server; 3653 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3654 3655 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3656 if (framesReq > framesReady) { 3657 framesReq = framesReady; 3658 } 3659 if (s + framesReq > bufferEnd) { 3660 framesReq = bufferEnd - s; 3661 } 3662 3663 buffer->raw = getBuffer(s, framesReq); 3664 if (buffer->raw == NULL) goto getNextBuffer_exit; 3665 3666 buffer->frameCount = framesReq; 3667 return NO_ERROR; 3668 } 3669 3670getNextBuffer_exit: 3671 buffer->raw = NULL; 3672 buffer->frameCount = 0; 3673 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3674 return NOT_ENOUGH_DATA; 3675} 3676 3677uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3678 return mCblk->framesReady(); 3679} 3680 3681bool AudioFlinger::PlaybackThread::Track::isReady() const { 3682 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3683 3684 if (framesReady() >= mCblk->frameCount || 3685 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3686 mFillingUpStatus = FS_FILLED; 3687 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3688 return true; 3689 } 3690 return false; 3691} 3692 3693status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid, 3694 AudioSystem::sync_event_t event, 3695 int triggerSession) 3696{ 3697 status_t status = NO_ERROR; 3698 ALOGV("start(%d), calling pid %d session %d tid %d", 3699 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3700 sp<ThreadBase> thread = mThread.promote(); 3701 if (thread != 0) { 3702 Mutex::Autolock _l(thread->mLock); 3703 track_state state = mState; 3704 // here the track could be either new, or restarted 3705 // in both cases "unstop" the track 3706 if (mState == PAUSED) { 3707 mState = TrackBase::RESUMING; 3708 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3709 } else { 3710 mState = TrackBase::ACTIVE; 3711 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3712 } 3713 3714 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3715 thread->mLock.unlock(); 3716 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3717 thread->mLock.lock(); 3718 3719#ifdef ADD_BATTERY_DATA 3720 // to track the speaker usage 3721 if (status == NO_ERROR) { 3722 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3723 } 3724#endif 3725 } 3726 if (status == NO_ERROR) { 3727 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3728 playbackThread->addTrack_l(this); 3729 } else { 3730 mState = state; 3731 } 3732 } else { 3733 status = BAD_VALUE; 3734 } 3735 return status; 3736} 3737 3738void AudioFlinger::PlaybackThread::Track::stop() 3739{ 3740 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3741 sp<ThreadBase> thread = mThread.promote(); 3742 if (thread != 0) { 3743 Mutex::Autolock _l(thread->mLock); 3744 track_state state = mState; 3745 if (mState > STOPPED) { 3746 mState = STOPPED; 3747 // If the track is not active (PAUSED and buffers full), flush buffers 3748 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3749 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3750 reset(); 3751 } 3752 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3753 } 3754 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3755 thread->mLock.unlock(); 3756 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3757 thread->mLock.lock(); 3758 3759#ifdef ADD_BATTERY_DATA 3760 // to track the speaker usage 3761 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3762#endif 3763 } 3764 } 3765} 3766 3767void AudioFlinger::PlaybackThread::Track::pause() 3768{ 3769 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3770 sp<ThreadBase> thread = mThread.promote(); 3771 if (thread != 0) { 3772 Mutex::Autolock _l(thread->mLock); 3773 if (mState == ACTIVE || mState == RESUMING) { 3774 mState = PAUSING; 3775 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3776 if (!isOutputTrack()) { 3777 thread->mLock.unlock(); 3778 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3779 thread->mLock.lock(); 3780 3781#ifdef ADD_BATTERY_DATA 3782 // to track the speaker usage 3783 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3784#endif 3785 } 3786 } 3787 } 3788} 3789 3790void AudioFlinger::PlaybackThread::Track::flush() 3791{ 3792 ALOGV("flush(%d)", mName); 3793 sp<ThreadBase> thread = mThread.promote(); 3794 if (thread != 0) { 3795 Mutex::Autolock _l(thread->mLock); 3796 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3797 return; 3798 } 3799 // No point remaining in PAUSED state after a flush => go to 3800 // STOPPED state 3801 mState = STOPPED; 3802 3803 // do not reset the track if it is still in the process of being stopped or paused. 3804 // this will be done by prepareTracks_l() when the track is stopped. 3805 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3806 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3807 reset(); 3808 } 3809 } 3810} 3811 3812void AudioFlinger::PlaybackThread::Track::reset() 3813{ 3814 // Do not reset twice to avoid discarding data written just after a flush and before 3815 // the audioflinger thread detects the track is stopped. 3816 if (!mResetDone) { 3817 TrackBase::reset(); 3818 // Force underrun condition to avoid false underrun callback until first data is 3819 // written to buffer 3820 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3821 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3822 mFillingUpStatus = FS_FILLING; 3823 mResetDone = true; 3824 mPresentationCompleteFrames = 0; 3825 } 3826} 3827 3828void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3829{ 3830 mMute = muted; 3831} 3832 3833status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3834{ 3835 status_t status = DEAD_OBJECT; 3836 sp<ThreadBase> thread = mThread.promote(); 3837 if (thread != 0) { 3838 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3839 status = playbackThread->attachAuxEffect(this, EffectId); 3840 } 3841 return status; 3842} 3843 3844void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3845{ 3846 mAuxEffectId = EffectId; 3847 mAuxBuffer = buffer; 3848} 3849 3850bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 3851 size_t audioHalFrames) 3852{ 3853 // a track is considered presented when the total number of frames written to audio HAL 3854 // corresponds to the number of frames written when presentationComplete() is called for the 3855 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 3856 if (mPresentationCompleteFrames == 0) { 3857 mPresentationCompleteFrames = framesWritten + audioHalFrames; 3858 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 3859 mPresentationCompleteFrames, audioHalFrames); 3860 } 3861 if (framesWritten >= mPresentationCompleteFrames) { 3862 ALOGV("presentationComplete() session %d complete: framesWritten %d", 3863 mSessionId, framesWritten); 3864 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 3865 mPresentationCompleteFrames = 0; 3866 return true; 3867 } 3868 return false; 3869} 3870 3871void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 3872{ 3873 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 3874 if (mSyncEvents[i]->type() == type) { 3875 mSyncEvents[i]->trigger(); 3876 mSyncEvents.removeAt(i); 3877 i--; 3878 } 3879 } 3880} 3881 3882 3883// timed audio tracks 3884 3885sp<AudioFlinger::PlaybackThread::TimedTrack> 3886AudioFlinger::PlaybackThread::TimedTrack::create( 3887 PlaybackThread *thread, 3888 const sp<Client>& client, 3889 audio_stream_type_t streamType, 3890 uint32_t sampleRate, 3891 audio_format_t format, 3892 uint32_t channelMask, 3893 int frameCount, 3894 const sp<IMemory>& sharedBuffer, 3895 int sessionId) { 3896 if (!client->reserveTimedTrack()) 3897 return NULL; 3898 3899 return new TimedTrack( 3900 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3901 sharedBuffer, sessionId); 3902} 3903 3904AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3905 PlaybackThread *thread, 3906 const sp<Client>& client, 3907 audio_stream_type_t streamType, 3908 uint32_t sampleRate, 3909 audio_format_t format, 3910 uint32_t channelMask, 3911 int frameCount, 3912 const sp<IMemory>& sharedBuffer, 3913 int sessionId) 3914 : Track(thread, client, streamType, sampleRate, format, channelMask, 3915 frameCount, sharedBuffer, sessionId), 3916 mTimedSilenceBuffer(NULL), 3917 mTimedSilenceBufferSize(0), 3918 mTimedAudioOutputOnTime(false), 3919 mMediaTimeTransformValid(false) 3920{ 3921 LocalClock lc; 3922 mLocalTimeFreq = lc.getLocalFreq(); 3923 3924 mLocalTimeToSampleTransform.a_zero = 0; 3925 mLocalTimeToSampleTransform.b_zero = 0; 3926 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3927 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3928 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3929 &mLocalTimeToSampleTransform.a_to_b_denom); 3930} 3931 3932AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3933 mClient->releaseTimedTrack(); 3934 delete [] mTimedSilenceBuffer; 3935} 3936 3937status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3938 size_t size, sp<IMemory>* buffer) { 3939 3940 Mutex::Autolock _l(mTimedBufferQueueLock); 3941 3942 trimTimedBufferQueue_l(); 3943 3944 // lazily initialize the shared memory heap for timed buffers 3945 if (mTimedMemoryDealer == NULL) { 3946 const int kTimedBufferHeapSize = 512 << 10; 3947 3948 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3949 "AudioFlingerTimed"); 3950 if (mTimedMemoryDealer == NULL) 3951 return NO_MEMORY; 3952 } 3953 3954 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3955 if (newBuffer == NULL) { 3956 newBuffer = mTimedMemoryDealer->allocate(size); 3957 if (newBuffer == NULL) 3958 return NO_MEMORY; 3959 } 3960 3961 *buffer = newBuffer; 3962 return NO_ERROR; 3963} 3964 3965// caller must hold mTimedBufferQueueLock 3966void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3967 int64_t mediaTimeNow; 3968 { 3969 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3970 if (!mMediaTimeTransformValid) 3971 return; 3972 3973 int64_t targetTimeNow; 3974 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3975 ? mCCHelper.getCommonTime(&targetTimeNow) 3976 : mCCHelper.getLocalTime(&targetTimeNow); 3977 3978 if (OK != res) 3979 return; 3980 3981 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3982 &mediaTimeNow)) { 3983 return; 3984 } 3985 } 3986 3987 size_t trimIndex; 3988 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3989 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3990 break; 3991 } 3992 3993 if (trimIndex) { 3994 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3995 } 3996} 3997 3998status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3999 const sp<IMemory>& buffer, int64_t pts) { 4000 4001 { 4002 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4003 if (!mMediaTimeTransformValid) 4004 return INVALID_OPERATION; 4005 } 4006 4007 Mutex::Autolock _l(mTimedBufferQueueLock); 4008 4009 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4010 4011 return NO_ERROR; 4012} 4013 4014status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4015 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4016 4017 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 4018 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4019 target); 4020 4021 if (!(target == TimedAudioTrack::LOCAL_TIME || 4022 target == TimedAudioTrack::COMMON_TIME)) { 4023 return BAD_VALUE; 4024 } 4025 4026 Mutex::Autolock lock(mMediaTimeTransformLock); 4027 mMediaTimeTransform = xform; 4028 mMediaTimeTransformTarget = target; 4029 mMediaTimeTransformValid = true; 4030 4031 return NO_ERROR; 4032} 4033 4034#define min(a, b) ((a) < (b) ? (a) : (b)) 4035 4036// implementation of getNextBuffer for tracks whose buffers have timestamps 4037status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4038 AudioBufferProvider::Buffer* buffer, int64_t pts) 4039{ 4040 if (pts == AudioBufferProvider::kInvalidPTS) { 4041 buffer->raw = 0; 4042 buffer->frameCount = 0; 4043 return INVALID_OPERATION; 4044 } 4045 4046 Mutex::Autolock _l(mTimedBufferQueueLock); 4047 4048 while (true) { 4049 4050 // if we have no timed buffers, then fail 4051 if (mTimedBufferQueue.isEmpty()) { 4052 buffer->raw = 0; 4053 buffer->frameCount = 0; 4054 return NOT_ENOUGH_DATA; 4055 } 4056 4057 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4058 4059 // calculate the PTS of the head of the timed buffer queue expressed in 4060 // local time 4061 int64_t headLocalPTS; 4062 { 4063 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4064 4065 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4066 4067 if (mMediaTimeTransform.a_to_b_denom == 0) { 4068 // the transform represents a pause, so yield silence 4069 timedYieldSilence(buffer->frameCount, buffer); 4070 return NO_ERROR; 4071 } 4072 4073 int64_t transformedPTS; 4074 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4075 &transformedPTS)) { 4076 // the transform failed. this shouldn't happen, but if it does 4077 // then just drop this buffer 4078 ALOGW("timedGetNextBuffer transform failed"); 4079 buffer->raw = 0; 4080 buffer->frameCount = 0; 4081 mTimedBufferQueue.removeAt(0); 4082 return NO_ERROR; 4083 } 4084 4085 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4086 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4087 &headLocalPTS)) { 4088 buffer->raw = 0; 4089 buffer->frameCount = 0; 4090 return INVALID_OPERATION; 4091 } 4092 } else { 4093 headLocalPTS = transformedPTS; 4094 } 4095 } 4096 4097 // adjust the head buffer's PTS to reflect the portion of the head buffer 4098 // that has already been consumed 4099 int64_t effectivePTS = headLocalPTS + 4100 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4101 4102 // Calculate the delta in samples between the head of the input buffer 4103 // queue and the start of the next output buffer that will be written. 4104 // If the transformation fails because of over or underflow, it means 4105 // that the sample's position in the output stream is so far out of 4106 // whack that it should just be dropped. 4107 int64_t sampleDelta; 4108 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4109 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4110 mTimedBufferQueue.removeAt(0); 4111 continue; 4112 } 4113 if (!mLocalTimeToSampleTransform.doForwardTransform( 4114 (effectivePTS - pts) << 32, &sampleDelta)) { 4115 ALOGV("*** too late during sample rate transform: dropped buffer"); 4116 mTimedBufferQueue.removeAt(0); 4117 continue; 4118 } 4119 4120 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4121 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4122 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4123 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4124 4125 // if the delta between the ideal placement for the next input sample and 4126 // the current output position is within this threshold, then we will 4127 // concatenate the next input samples to the previous output 4128 const int64_t kSampleContinuityThreshold = 4129 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4130 4131 // if this is the first buffer of audio that we're emitting from this track 4132 // then it should be almost exactly on time. 4133 const int64_t kSampleStartupThreshold = 1LL << 32; 4134 4135 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4136 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4137 // the next input is close enough to being on time, so concatenate it 4138 // with the last output 4139 timedYieldSamples(buffer); 4140 4141 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4142 return NO_ERROR; 4143 } else if (sampleDelta > 0) { 4144 // the gap between the current output position and the proper start of 4145 // the next input sample is too big, so fill it with silence 4146 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4147 4148 timedYieldSilence(framesUntilNextInput, buffer); 4149 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4150 return NO_ERROR; 4151 } else { 4152 // the next input sample is late 4153 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4154 size_t onTimeSamplePosition = 4155 head.position() + lateFrames * mCblk->frameSize; 4156 4157 if (onTimeSamplePosition > head.buffer()->size()) { 4158 // all the remaining samples in the head are too late, so 4159 // drop it and move on 4160 ALOGV("*** too late: dropped buffer"); 4161 mTimedBufferQueue.removeAt(0); 4162 continue; 4163 } else { 4164 // skip over the late samples 4165 head.setPosition(onTimeSamplePosition); 4166 4167 // yield the available samples 4168 timedYieldSamples(buffer); 4169 4170 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4171 return NO_ERROR; 4172 } 4173 } 4174 } 4175} 4176 4177// Yield samples from the timed buffer queue head up to the given output 4178// buffer's capacity. 4179// 4180// Caller must hold mTimedBufferQueueLock 4181void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4182 AudioBufferProvider::Buffer* buffer) { 4183 4184 const TimedBuffer& head = mTimedBufferQueue[0]; 4185 4186 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4187 head.position()); 4188 4189 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4190 mCblk->frameSize); 4191 size_t framesRequested = buffer->frameCount; 4192 buffer->frameCount = min(framesLeftInHead, framesRequested); 4193 4194 mTimedAudioOutputOnTime = true; 4195} 4196 4197// Yield samples of silence up to the given output buffer's capacity 4198// 4199// Caller must hold mTimedBufferQueueLock 4200void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4201 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4202 4203 // lazily allocate a buffer filled with silence 4204 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4205 delete [] mTimedSilenceBuffer; 4206 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4207 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4208 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4209 } 4210 4211 buffer->raw = mTimedSilenceBuffer; 4212 size_t framesRequested = buffer->frameCount; 4213 buffer->frameCount = min(numFrames, framesRequested); 4214 4215 mTimedAudioOutputOnTime = false; 4216} 4217 4218// AudioBufferProvider interface 4219void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4220 AudioBufferProvider::Buffer* buffer) { 4221 4222 Mutex::Autolock _l(mTimedBufferQueueLock); 4223 4224 // If the buffer which was just released is part of the buffer at the head 4225 // of the queue, be sure to update the amt of the buffer which has been 4226 // consumed. If the buffer being returned is not part of the head of the 4227 // queue, its either because the buffer is part of the silence buffer, or 4228 // because the head of the timed queue was trimmed after the mixer called 4229 // getNextBuffer but before the mixer called releaseBuffer. 4230 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4231 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4232 4233 void* start = head.buffer()->pointer(); 4234 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4235 4236 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4237 head.setPosition(head.position() + 4238 (buffer->frameCount * mCblk->frameSize)); 4239 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4240 mTimedBufferQueue.removeAt(0); 4241 } 4242 } 4243 } 4244 4245 buffer->raw = 0; 4246 buffer->frameCount = 0; 4247} 4248 4249uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4250 Mutex::Autolock _l(mTimedBufferQueueLock); 4251 4252 uint32_t frames = 0; 4253 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4254 const TimedBuffer& tb = mTimedBufferQueue[i]; 4255 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4256 } 4257 4258 return frames; 4259} 4260 4261AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4262 : mPTS(0), mPosition(0) {} 4263 4264AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4265 const sp<IMemory>& buffer, int64_t pts) 4266 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4267 4268// ---------------------------------------------------------------------------- 4269 4270// RecordTrack constructor must be called with AudioFlinger::mLock held 4271AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4272 RecordThread *thread, 4273 const sp<Client>& client, 4274 uint32_t sampleRate, 4275 audio_format_t format, 4276 uint32_t channelMask, 4277 int frameCount, 4278 int sessionId) 4279 : TrackBase(thread, client, sampleRate, format, 4280 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4281 mOverflow(false) 4282{ 4283 if (mCblk != NULL) { 4284 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4285 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4286 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4287 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4288 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4289 } else { 4290 mCblk->frameSize = sizeof(int8_t); 4291 } 4292 } 4293} 4294 4295AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4296{ 4297 sp<ThreadBase> thread = mThread.promote(); 4298 if (thread != 0) { 4299 AudioSystem::releaseInput(thread->id()); 4300 } 4301} 4302 4303// AudioBufferProvider interface 4304status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4305{ 4306 audio_track_cblk_t* cblk = this->cblk(); 4307 uint32_t framesAvail; 4308 uint32_t framesReq = buffer->frameCount; 4309 4310 // Check if last stepServer failed, try to step now 4311 if (mStepServerFailed) { 4312 if (!step()) goto getNextBuffer_exit; 4313 ALOGV("stepServer recovered"); 4314 mStepServerFailed = false; 4315 } 4316 4317 framesAvail = cblk->framesAvailable_l(); 4318 4319 if (CC_LIKELY(framesAvail)) { 4320 uint32_t s = cblk->server; 4321 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4322 4323 if (framesReq > framesAvail) { 4324 framesReq = framesAvail; 4325 } 4326 if (s + framesReq > bufferEnd) { 4327 framesReq = bufferEnd - s; 4328 } 4329 4330 buffer->raw = getBuffer(s, framesReq); 4331 if (buffer->raw == NULL) goto getNextBuffer_exit; 4332 4333 buffer->frameCount = framesReq; 4334 return NO_ERROR; 4335 } 4336 4337getNextBuffer_exit: 4338 buffer->raw = NULL; 4339 buffer->frameCount = 0; 4340 return NOT_ENOUGH_DATA; 4341} 4342 4343status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid, 4344 AudioSystem::sync_event_t event, 4345 int triggerSession) 4346{ 4347 sp<ThreadBase> thread = mThread.promote(); 4348 if (thread != 0) { 4349 RecordThread *recordThread = (RecordThread *)thread.get(); 4350 return recordThread->start(this, tid, event, triggerSession); 4351 } else { 4352 return BAD_VALUE; 4353 } 4354} 4355 4356void AudioFlinger::RecordThread::RecordTrack::stop() 4357{ 4358 sp<ThreadBase> thread = mThread.promote(); 4359 if (thread != 0) { 4360 RecordThread *recordThread = (RecordThread *)thread.get(); 4361 recordThread->stop(this); 4362 TrackBase::reset(); 4363 // Force overrun condition to avoid false overrun callback until first data is 4364 // read from buffer 4365 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4366 } 4367} 4368 4369void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4370{ 4371 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4372 (mClient == 0) ? getpid_cached : mClient->pid(), 4373 mFormat, 4374 mChannelMask, 4375 mSessionId, 4376 mFrameCount, 4377 mState, 4378 mCblk->sampleRate, 4379 mCblk->server, 4380 mCblk->user); 4381} 4382 4383 4384// ---------------------------------------------------------------------------- 4385 4386AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4387 PlaybackThread *playbackThread, 4388 DuplicatingThread *sourceThread, 4389 uint32_t sampleRate, 4390 audio_format_t format, 4391 uint32_t channelMask, 4392 int frameCount) 4393 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4394 mActive(false), mSourceThread(sourceThread) 4395{ 4396 4397 if (mCblk != NULL) { 4398 mCblk->flags |= CBLK_DIRECTION_OUT; 4399 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4400 mOutBuffer.frameCount = 0; 4401 playbackThread->mTracks.add(this); 4402 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4403 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4404 mCblk, mBuffer, mCblk->buffers, 4405 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4406 } else { 4407 ALOGW("Error creating output track on thread %p", playbackThread); 4408 } 4409} 4410 4411AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4412{ 4413 clearBufferQueue(); 4414} 4415 4416status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid, 4417 AudioSystem::sync_event_t event, 4418 int triggerSession) 4419{ 4420 status_t status = Track::start(tid, event, triggerSession); 4421 if (status != NO_ERROR) { 4422 return status; 4423 } 4424 4425 mActive = true; 4426 mRetryCount = 127; 4427 return status; 4428} 4429 4430void AudioFlinger::PlaybackThread::OutputTrack::stop() 4431{ 4432 Track::stop(); 4433 clearBufferQueue(); 4434 mOutBuffer.frameCount = 0; 4435 mActive = false; 4436} 4437 4438bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4439{ 4440 Buffer *pInBuffer; 4441 Buffer inBuffer; 4442 uint32_t channelCount = mChannelCount; 4443 bool outputBufferFull = false; 4444 inBuffer.frameCount = frames; 4445 inBuffer.i16 = data; 4446 4447 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4448 4449 if (!mActive && frames != 0) { 4450 start(0); 4451 sp<ThreadBase> thread = mThread.promote(); 4452 if (thread != 0) { 4453 MixerThread *mixerThread = (MixerThread *)thread.get(); 4454 if (mCblk->frameCount > frames){ 4455 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4456 uint32_t startFrames = (mCblk->frameCount - frames); 4457 pInBuffer = new Buffer; 4458 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4459 pInBuffer->frameCount = startFrames; 4460 pInBuffer->i16 = pInBuffer->mBuffer; 4461 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4462 mBufferQueue.add(pInBuffer); 4463 } else { 4464 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4465 } 4466 } 4467 } 4468 } 4469 4470 while (waitTimeLeftMs) { 4471 // First write pending buffers, then new data 4472 if (mBufferQueue.size()) { 4473 pInBuffer = mBufferQueue.itemAt(0); 4474 } else { 4475 pInBuffer = &inBuffer; 4476 } 4477 4478 if (pInBuffer->frameCount == 0) { 4479 break; 4480 } 4481 4482 if (mOutBuffer.frameCount == 0) { 4483 mOutBuffer.frameCount = pInBuffer->frameCount; 4484 nsecs_t startTime = systemTime(); 4485 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4486 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4487 outputBufferFull = true; 4488 break; 4489 } 4490 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4491 if (waitTimeLeftMs >= waitTimeMs) { 4492 waitTimeLeftMs -= waitTimeMs; 4493 } else { 4494 waitTimeLeftMs = 0; 4495 } 4496 } 4497 4498 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4499 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4500 mCblk->stepUser(outFrames); 4501 pInBuffer->frameCount -= outFrames; 4502 pInBuffer->i16 += outFrames * channelCount; 4503 mOutBuffer.frameCount -= outFrames; 4504 mOutBuffer.i16 += outFrames * channelCount; 4505 4506 if (pInBuffer->frameCount == 0) { 4507 if (mBufferQueue.size()) { 4508 mBufferQueue.removeAt(0); 4509 delete [] pInBuffer->mBuffer; 4510 delete pInBuffer; 4511 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4512 } else { 4513 break; 4514 } 4515 } 4516 } 4517 4518 // If we could not write all frames, allocate a buffer and queue it for next time. 4519 if (inBuffer.frameCount) { 4520 sp<ThreadBase> thread = mThread.promote(); 4521 if (thread != 0 && !thread->standby()) { 4522 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4523 pInBuffer = new Buffer; 4524 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4525 pInBuffer->frameCount = inBuffer.frameCount; 4526 pInBuffer->i16 = pInBuffer->mBuffer; 4527 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4528 mBufferQueue.add(pInBuffer); 4529 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4530 } else { 4531 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4532 } 4533 } 4534 } 4535 4536 // Calling write() with a 0 length buffer, means that no more data will be written: 4537 // If no more buffers are pending, fill output track buffer to make sure it is started 4538 // by output mixer. 4539 if (frames == 0 && mBufferQueue.size() == 0) { 4540 if (mCblk->user < mCblk->frameCount) { 4541 frames = mCblk->frameCount - mCblk->user; 4542 pInBuffer = new Buffer; 4543 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4544 pInBuffer->frameCount = frames; 4545 pInBuffer->i16 = pInBuffer->mBuffer; 4546 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4547 mBufferQueue.add(pInBuffer); 4548 } else if (mActive) { 4549 stop(); 4550 } 4551 } 4552 4553 return outputBufferFull; 4554} 4555 4556status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4557{ 4558 int active; 4559 status_t result; 4560 audio_track_cblk_t* cblk = mCblk; 4561 uint32_t framesReq = buffer->frameCount; 4562 4563// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4564 buffer->frameCount = 0; 4565 4566 uint32_t framesAvail = cblk->framesAvailable(); 4567 4568 4569 if (framesAvail == 0) { 4570 Mutex::Autolock _l(cblk->lock); 4571 goto start_loop_here; 4572 while (framesAvail == 0) { 4573 active = mActive; 4574 if (CC_UNLIKELY(!active)) { 4575 ALOGV("Not active and NO_MORE_BUFFERS"); 4576 return NO_MORE_BUFFERS; 4577 } 4578 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4579 if (result != NO_ERROR) { 4580 return NO_MORE_BUFFERS; 4581 } 4582 // read the server count again 4583 start_loop_here: 4584 framesAvail = cblk->framesAvailable_l(); 4585 } 4586 } 4587 4588// if (framesAvail < framesReq) { 4589// return NO_MORE_BUFFERS; 4590// } 4591 4592 if (framesReq > framesAvail) { 4593 framesReq = framesAvail; 4594 } 4595 4596 uint32_t u = cblk->user; 4597 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4598 4599 if (u + framesReq > bufferEnd) { 4600 framesReq = bufferEnd - u; 4601 } 4602 4603 buffer->frameCount = framesReq; 4604 buffer->raw = (void *)cblk->buffer(u); 4605 return NO_ERROR; 4606} 4607 4608 4609void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4610{ 4611 size_t size = mBufferQueue.size(); 4612 4613 for (size_t i = 0; i < size; i++) { 4614 Buffer *pBuffer = mBufferQueue.itemAt(i); 4615 delete [] pBuffer->mBuffer; 4616 delete pBuffer; 4617 } 4618 mBufferQueue.clear(); 4619} 4620 4621// ---------------------------------------------------------------------------- 4622 4623AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4624 : RefBase(), 4625 mAudioFlinger(audioFlinger), 4626 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4627 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4628 mPid(pid), 4629 mTimedTrackCount(0) 4630{ 4631 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4632} 4633 4634// Client destructor must be called with AudioFlinger::mLock held 4635AudioFlinger::Client::~Client() 4636{ 4637 mAudioFlinger->removeClient_l(mPid); 4638} 4639 4640sp<MemoryDealer> AudioFlinger::Client::heap() const 4641{ 4642 return mMemoryDealer; 4643} 4644 4645// Reserve one of the limited slots for a timed audio track associated 4646// with this client 4647bool AudioFlinger::Client::reserveTimedTrack() 4648{ 4649 const int kMaxTimedTracksPerClient = 4; 4650 4651 Mutex::Autolock _l(mTimedTrackLock); 4652 4653 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4654 ALOGW("can not create timed track - pid %d has exceeded the limit", 4655 mPid); 4656 return false; 4657 } 4658 4659 mTimedTrackCount++; 4660 return true; 4661} 4662 4663// Release a slot for a timed audio track 4664void AudioFlinger::Client::releaseTimedTrack() 4665{ 4666 Mutex::Autolock _l(mTimedTrackLock); 4667 mTimedTrackCount--; 4668} 4669 4670// ---------------------------------------------------------------------------- 4671 4672AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4673 const sp<IAudioFlingerClient>& client, 4674 pid_t pid) 4675 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4676{ 4677} 4678 4679AudioFlinger::NotificationClient::~NotificationClient() 4680{ 4681} 4682 4683void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4684{ 4685 sp<NotificationClient> keep(this); 4686 mAudioFlinger->removeNotificationClient(mPid); 4687} 4688 4689// ---------------------------------------------------------------------------- 4690 4691AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4692 : BnAudioTrack(), 4693 mTrack(track) 4694{ 4695} 4696 4697AudioFlinger::TrackHandle::~TrackHandle() { 4698 // just stop the track on deletion, associated resources 4699 // will be freed from the main thread once all pending buffers have 4700 // been played. Unless it's not in the active track list, in which 4701 // case we free everything now... 4702 mTrack->destroy(); 4703} 4704 4705sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4706 return mTrack->getCblk(); 4707} 4708 4709status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4710 return mTrack->start(tid); 4711} 4712 4713void AudioFlinger::TrackHandle::stop() { 4714 mTrack->stop(); 4715} 4716 4717void AudioFlinger::TrackHandle::flush() { 4718 mTrack->flush(); 4719} 4720 4721void AudioFlinger::TrackHandle::mute(bool e) { 4722 mTrack->mute(e); 4723} 4724 4725void AudioFlinger::TrackHandle::pause() { 4726 mTrack->pause(); 4727} 4728 4729status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4730{ 4731 return mTrack->attachAuxEffect(EffectId); 4732} 4733 4734status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4735 sp<IMemory>* buffer) { 4736 if (!mTrack->isTimedTrack()) 4737 return INVALID_OPERATION; 4738 4739 PlaybackThread::TimedTrack* tt = 4740 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4741 return tt->allocateTimedBuffer(size, buffer); 4742} 4743 4744status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4745 int64_t pts) { 4746 if (!mTrack->isTimedTrack()) 4747 return INVALID_OPERATION; 4748 4749 PlaybackThread::TimedTrack* tt = 4750 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4751 return tt->queueTimedBuffer(buffer, pts); 4752} 4753 4754status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4755 const LinearTransform& xform, int target) { 4756 4757 if (!mTrack->isTimedTrack()) 4758 return INVALID_OPERATION; 4759 4760 PlaybackThread::TimedTrack* tt = 4761 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4762 return tt->setMediaTimeTransform( 4763 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4764} 4765 4766status_t AudioFlinger::TrackHandle::onTransact( 4767 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4768{ 4769 return BnAudioTrack::onTransact(code, data, reply, flags); 4770} 4771 4772// ---------------------------------------------------------------------------- 4773 4774sp<IAudioRecord> AudioFlinger::openRecord( 4775 pid_t pid, 4776 audio_io_handle_t input, 4777 uint32_t sampleRate, 4778 audio_format_t format, 4779 uint32_t channelMask, 4780 int frameCount, 4781 IAudioFlinger::track_flags_t flags, 4782 int *sessionId, 4783 status_t *status) 4784{ 4785 sp<RecordThread::RecordTrack> recordTrack; 4786 sp<RecordHandle> recordHandle; 4787 sp<Client> client; 4788 status_t lStatus; 4789 RecordThread *thread; 4790 size_t inFrameCount; 4791 int lSessionId; 4792 4793 // check calling permissions 4794 if (!recordingAllowed()) { 4795 lStatus = PERMISSION_DENIED; 4796 goto Exit; 4797 } 4798 4799 // add client to list 4800 { // scope for mLock 4801 Mutex::Autolock _l(mLock); 4802 thread = checkRecordThread_l(input); 4803 if (thread == NULL) { 4804 lStatus = BAD_VALUE; 4805 goto Exit; 4806 } 4807 4808 client = registerPid_l(pid); 4809 4810 // If no audio session id is provided, create one here 4811 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4812 lSessionId = *sessionId; 4813 } else { 4814 lSessionId = nextUniqueId(); 4815 if (sessionId != NULL) { 4816 *sessionId = lSessionId; 4817 } 4818 } 4819 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4820 recordTrack = thread->createRecordTrack_l(client, 4821 sampleRate, 4822 format, 4823 channelMask, 4824 frameCount, 4825 lSessionId, 4826 &lStatus); 4827 } 4828 if (lStatus != NO_ERROR) { 4829 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4830 // destructor is called by the TrackBase destructor with mLock held 4831 client.clear(); 4832 recordTrack.clear(); 4833 goto Exit; 4834 } 4835 4836 // return to handle to client 4837 recordHandle = new RecordHandle(recordTrack); 4838 lStatus = NO_ERROR; 4839 4840Exit: 4841 if (status) { 4842 *status = lStatus; 4843 } 4844 return recordHandle; 4845} 4846 4847// ---------------------------------------------------------------------------- 4848 4849AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4850 : BnAudioRecord(), 4851 mRecordTrack(recordTrack) 4852{ 4853} 4854 4855AudioFlinger::RecordHandle::~RecordHandle() { 4856 stop(); 4857} 4858 4859sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4860 return mRecordTrack->getCblk(); 4861} 4862 4863status_t AudioFlinger::RecordHandle::start(pid_t tid, int event, int triggerSession) { 4864 ALOGV("RecordHandle::start()"); 4865 return mRecordTrack->start(tid, (AudioSystem::sync_event_t)event, triggerSession); 4866} 4867 4868void AudioFlinger::RecordHandle::stop() { 4869 ALOGV("RecordHandle::stop()"); 4870 mRecordTrack->stop(); 4871} 4872 4873status_t AudioFlinger::RecordHandle::onTransact( 4874 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4875{ 4876 return BnAudioRecord::onTransact(code, data, reply, flags); 4877} 4878 4879// ---------------------------------------------------------------------------- 4880 4881AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4882 AudioStreamIn *input, 4883 uint32_t sampleRate, 4884 uint32_t channels, 4885 audio_io_handle_t id, 4886 uint32_t device) : 4887 ThreadBase(audioFlinger, id, device, RECORD), 4888 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4889 // mRsmpInIndex and mInputBytes set by readInputParameters() 4890 mReqChannelCount(popcount(channels)), 4891 mReqSampleRate(sampleRate) 4892 // mBytesRead is only meaningful while active, and so is cleared in start() 4893 // (but might be better to also clear here for dump?) 4894{ 4895 snprintf(mName, kNameLength, "AudioIn_%X", id); 4896 4897 readInputParameters(); 4898} 4899 4900 4901AudioFlinger::RecordThread::~RecordThread() 4902{ 4903 delete[] mRsmpInBuffer; 4904 delete mResampler; 4905 delete[] mRsmpOutBuffer; 4906} 4907 4908void AudioFlinger::RecordThread::onFirstRef() 4909{ 4910 run(mName, PRIORITY_URGENT_AUDIO); 4911} 4912 4913status_t AudioFlinger::RecordThread::readyToRun() 4914{ 4915 status_t status = initCheck(); 4916 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4917 return status; 4918} 4919 4920bool AudioFlinger::RecordThread::threadLoop() 4921{ 4922 AudioBufferProvider::Buffer buffer; 4923 sp<RecordTrack> activeTrack; 4924 Vector< sp<EffectChain> > effectChains; 4925 4926 nsecs_t lastWarning = 0; 4927 4928 acquireWakeLock(); 4929 4930 // start recording 4931 while (!exitPending()) { 4932 4933 processConfigEvents(); 4934 4935 { // scope for mLock 4936 Mutex::Autolock _l(mLock); 4937 checkForNewParameters_l(); 4938 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4939 if (!mStandby) { 4940 mInput->stream->common.standby(&mInput->stream->common); 4941 mStandby = true; 4942 } 4943 4944 if (exitPending()) break; 4945 4946 releaseWakeLock_l(); 4947 ALOGV("RecordThread: loop stopping"); 4948 // go to sleep 4949 mWaitWorkCV.wait(mLock); 4950 ALOGV("RecordThread: loop starting"); 4951 acquireWakeLock_l(); 4952 continue; 4953 } 4954 if (mActiveTrack != 0) { 4955 if (mActiveTrack->mState == TrackBase::PAUSING) { 4956 if (!mStandby) { 4957 mInput->stream->common.standby(&mInput->stream->common); 4958 mStandby = true; 4959 } 4960 mActiveTrack.clear(); 4961 mStartStopCond.broadcast(); 4962 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4963 if (mReqChannelCount != mActiveTrack->channelCount()) { 4964 mActiveTrack.clear(); 4965 mStartStopCond.broadcast(); 4966 } else if (mBytesRead != 0) { 4967 // record start succeeds only if first read from audio input 4968 // succeeds 4969 if (mBytesRead > 0) { 4970 mActiveTrack->mState = TrackBase::ACTIVE; 4971 } else { 4972 mActiveTrack.clear(); 4973 } 4974 mStartStopCond.broadcast(); 4975 } 4976 mStandby = false; 4977 } 4978 } 4979 lockEffectChains_l(effectChains); 4980 } 4981 4982 if (mActiveTrack != 0) { 4983 if (mActiveTrack->mState != TrackBase::ACTIVE && 4984 mActiveTrack->mState != TrackBase::RESUMING) { 4985 unlockEffectChains(effectChains); 4986 usleep(kRecordThreadSleepUs); 4987 continue; 4988 } 4989 for (size_t i = 0; i < effectChains.size(); i ++) { 4990 effectChains[i]->process_l(); 4991 } 4992 4993 buffer.frameCount = mFrameCount; 4994 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4995 size_t framesOut = buffer.frameCount; 4996 if (mResampler == NULL) { 4997 // no resampling 4998 while (framesOut) { 4999 size_t framesIn = mFrameCount - mRsmpInIndex; 5000 if (framesIn) { 5001 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5002 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5003 if (framesIn > framesOut) 5004 framesIn = framesOut; 5005 mRsmpInIndex += framesIn; 5006 framesOut -= framesIn; 5007 if ((int)mChannelCount == mReqChannelCount || 5008 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5009 memcpy(dst, src, framesIn * mFrameSize); 5010 } else { 5011 int16_t *src16 = (int16_t *)src; 5012 int16_t *dst16 = (int16_t *)dst; 5013 if (mChannelCount == 1) { 5014 while (framesIn--) { 5015 *dst16++ = *src16; 5016 *dst16++ = *src16++; 5017 } 5018 } else { 5019 while (framesIn--) { 5020 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5021 src16 += 2; 5022 } 5023 } 5024 } 5025 } 5026 if (framesOut && mFrameCount == mRsmpInIndex) { 5027 if (framesOut == mFrameCount && 5028 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5029 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5030 framesOut = 0; 5031 } else { 5032 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5033 mRsmpInIndex = 0; 5034 } 5035 if (mBytesRead < 0) { 5036 ALOGE("Error reading audio input"); 5037 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5038 // Force input into standby so that it tries to 5039 // recover at next read attempt 5040 mInput->stream->common.standby(&mInput->stream->common); 5041 usleep(kRecordThreadSleepUs); 5042 } 5043 mRsmpInIndex = mFrameCount; 5044 framesOut = 0; 5045 buffer.frameCount = 0; 5046 } 5047 } 5048 } 5049 } else { 5050 // resampling 5051 5052 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5053 // alter output frame count as if we were expecting stereo samples 5054 if (mChannelCount == 1 && mReqChannelCount == 1) { 5055 framesOut >>= 1; 5056 } 5057 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5058 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5059 // are 32 bit aligned which should be always true. 5060 if (mChannelCount == 2 && mReqChannelCount == 1) { 5061 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5062 // the resampler always outputs stereo samples: do post stereo to mono conversion 5063 int16_t *src = (int16_t *)mRsmpOutBuffer; 5064 int16_t *dst = buffer.i16; 5065 while (framesOut--) { 5066 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5067 src += 2; 5068 } 5069 } else { 5070 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5071 } 5072 5073 } 5074 if (mFramestoDrop == 0) { 5075 mActiveTrack->releaseBuffer(&buffer); 5076 } else { 5077 if (mFramestoDrop > 0) { 5078 mFramestoDrop -= buffer.frameCount; 5079 if (mFramestoDrop < 0) { 5080 mFramestoDrop = 0; 5081 } 5082 } 5083 } 5084 mActiveTrack->overflow(); 5085 } 5086 // client isn't retrieving buffers fast enough 5087 else { 5088 if (!mActiveTrack->setOverflow()) { 5089 nsecs_t now = systemTime(); 5090 if ((now - lastWarning) > kWarningThrottleNs) { 5091 ALOGW("RecordThread: buffer overflow"); 5092 lastWarning = now; 5093 } 5094 } 5095 // Release the processor for a while before asking for a new buffer. 5096 // This will give the application more chance to read from the buffer and 5097 // clear the overflow. 5098 usleep(kRecordThreadSleepUs); 5099 } 5100 } 5101 // enable changes in effect chain 5102 unlockEffectChains(effectChains); 5103 effectChains.clear(); 5104 } 5105 5106 if (!mStandby) { 5107 mInput->stream->common.standby(&mInput->stream->common); 5108 } 5109 mActiveTrack.clear(); 5110 5111 mStartStopCond.broadcast(); 5112 5113 releaseWakeLock(); 5114 5115 ALOGV("RecordThread %p exiting", this); 5116 return false; 5117} 5118 5119 5120sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5121 const sp<AudioFlinger::Client>& client, 5122 uint32_t sampleRate, 5123 audio_format_t format, 5124 int channelMask, 5125 int frameCount, 5126 int sessionId, 5127 status_t *status) 5128{ 5129 sp<RecordTrack> track; 5130 status_t lStatus; 5131 5132 lStatus = initCheck(); 5133 if (lStatus != NO_ERROR) { 5134 ALOGE("Audio driver not initialized."); 5135 goto Exit; 5136 } 5137 5138 { // scope for mLock 5139 Mutex::Autolock _l(mLock); 5140 5141 track = new RecordTrack(this, client, sampleRate, 5142 format, channelMask, frameCount, sessionId); 5143 5144 if (track->getCblk() == 0) { 5145 lStatus = NO_MEMORY; 5146 goto Exit; 5147 } 5148 5149 mTrack = track.get(); 5150 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5151 bool suspend = audio_is_bluetooth_sco_device( 5152 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5153 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5154 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5155 } 5156 lStatus = NO_ERROR; 5157 5158Exit: 5159 if (status) { 5160 *status = lStatus; 5161 } 5162 return track; 5163} 5164 5165status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5166 pid_t tid, AudioSystem::sync_event_t event, 5167 int triggerSession) 5168{ 5169 ALOGV("RecordThread::start tid=%d, event %d, triggerSession %d", tid, event, triggerSession); 5170 sp<ThreadBase> strongMe = this; 5171 status_t status = NO_ERROR; 5172 5173 if (event == AudioSystem::SYNC_EVENT_NONE) { 5174 mSyncStartEvent.clear(); 5175 mFramestoDrop = 0; 5176 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5177 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5178 triggerSession, 5179 recordTrack->sessionId(), 5180 syncStartEventCallback, 5181 this); 5182 mFramestoDrop = -1; 5183 } 5184 5185 { 5186 AutoMutex lock(mLock); 5187 if (mActiveTrack != 0) { 5188 if (recordTrack != mActiveTrack.get()) { 5189 status = -EBUSY; 5190 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5191 mActiveTrack->mState = TrackBase::ACTIVE; 5192 } 5193 return status; 5194 } 5195 5196 recordTrack->mState = TrackBase::IDLE; 5197 mActiveTrack = recordTrack; 5198 mLock.unlock(); 5199 status_t status = AudioSystem::startInput(mId); 5200 mLock.lock(); 5201 if (status != NO_ERROR) { 5202 mActiveTrack.clear(); 5203 clearSyncStartEvent(); 5204 return status; 5205 } 5206 mRsmpInIndex = mFrameCount; 5207 mBytesRead = 0; 5208 if (mResampler != NULL) { 5209 mResampler->reset(); 5210 } 5211 mActiveTrack->mState = TrackBase::RESUMING; 5212 // signal thread to start 5213 ALOGV("Signal record thread"); 5214 mWaitWorkCV.signal(); 5215 // do not wait for mStartStopCond if exiting 5216 if (exitPending()) { 5217 mActiveTrack.clear(); 5218 status = INVALID_OPERATION; 5219 goto startError; 5220 } 5221 mStartStopCond.wait(mLock); 5222 if (mActiveTrack == 0) { 5223 ALOGV("Record failed to start"); 5224 status = BAD_VALUE; 5225 goto startError; 5226 } 5227 ALOGV("Record started OK"); 5228 return status; 5229 } 5230startError: 5231 AudioSystem::stopInput(mId); 5232 clearSyncStartEvent(); 5233 return status; 5234} 5235 5236void AudioFlinger::RecordThread::clearSyncStartEvent() 5237{ 5238 if (mSyncStartEvent != 0) { 5239 mSyncStartEvent->cancel(); 5240 } 5241 mSyncStartEvent.clear(); 5242} 5243 5244void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5245{ 5246 sp<SyncEvent> strongEvent = event.promote(); 5247 5248 if (strongEvent != 0) { 5249 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5250 me->handleSyncStartEvent(strongEvent); 5251 } 5252} 5253 5254void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5255{ 5256 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5257 mActiveTrack.get(), 5258 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5259 event->listenerSession()); 5260 5261 if (mActiveTrack != 0 && 5262 event == mSyncStartEvent) { 5263 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5264 // from audio HAL 5265 mFramestoDrop = mFrameCount * 2; 5266 mSyncStartEvent.clear(); 5267 } 5268} 5269 5270void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5271 ALOGV("RecordThread::stop"); 5272 sp<ThreadBase> strongMe = this; 5273 { 5274 AutoMutex lock(mLock); 5275 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5276 mActiveTrack->mState = TrackBase::PAUSING; 5277 // do not wait for mStartStopCond if exiting 5278 if (exitPending()) { 5279 return; 5280 } 5281 mStartStopCond.wait(mLock); 5282 // if we have been restarted, recordTrack == mActiveTrack.get() here 5283 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5284 mLock.unlock(); 5285 AudioSystem::stopInput(mId); 5286 mLock.lock(); 5287 ALOGV("Record stopped OK"); 5288 } 5289 } 5290 } 5291} 5292 5293bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5294{ 5295 return false; 5296} 5297 5298status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5299{ 5300 if (!isValidSyncEvent(event)) { 5301 return BAD_VALUE; 5302 } 5303 5304 Mutex::Autolock _l(mLock); 5305 5306 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5307 mTrack->setSyncEvent(event); 5308 return NO_ERROR; 5309 } 5310 return NAME_NOT_FOUND; 5311} 5312 5313status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5314{ 5315 const size_t SIZE = 256; 5316 char buffer[SIZE]; 5317 String8 result; 5318 5319 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5320 result.append(buffer); 5321 5322 if (mActiveTrack != 0) { 5323 result.append("Active Track:\n"); 5324 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5325 mActiveTrack->dump(buffer, SIZE); 5326 result.append(buffer); 5327 5328 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5329 result.append(buffer); 5330 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5331 result.append(buffer); 5332 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5333 result.append(buffer); 5334 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5335 result.append(buffer); 5336 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5337 result.append(buffer); 5338 5339 5340 } else { 5341 result.append("No record client\n"); 5342 } 5343 write(fd, result.string(), result.size()); 5344 5345 dumpBase(fd, args); 5346 dumpEffectChains(fd, args); 5347 5348 return NO_ERROR; 5349} 5350 5351// AudioBufferProvider interface 5352status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5353{ 5354 size_t framesReq = buffer->frameCount; 5355 size_t framesReady = mFrameCount - mRsmpInIndex; 5356 int channelCount; 5357 5358 if (framesReady == 0) { 5359 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5360 if (mBytesRead < 0) { 5361 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5362 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5363 // Force input into standby so that it tries to 5364 // recover at next read attempt 5365 mInput->stream->common.standby(&mInput->stream->common); 5366 usleep(kRecordThreadSleepUs); 5367 } 5368 buffer->raw = NULL; 5369 buffer->frameCount = 0; 5370 return NOT_ENOUGH_DATA; 5371 } 5372 mRsmpInIndex = 0; 5373 framesReady = mFrameCount; 5374 } 5375 5376 if (framesReq > framesReady) { 5377 framesReq = framesReady; 5378 } 5379 5380 if (mChannelCount == 1 && mReqChannelCount == 2) { 5381 channelCount = 1; 5382 } else { 5383 channelCount = 2; 5384 } 5385 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5386 buffer->frameCount = framesReq; 5387 return NO_ERROR; 5388} 5389 5390// AudioBufferProvider interface 5391void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5392{ 5393 mRsmpInIndex += buffer->frameCount; 5394 buffer->frameCount = 0; 5395} 5396 5397bool AudioFlinger::RecordThread::checkForNewParameters_l() 5398{ 5399 bool reconfig = false; 5400 5401 while (!mNewParameters.isEmpty()) { 5402 status_t status = NO_ERROR; 5403 String8 keyValuePair = mNewParameters[0]; 5404 AudioParameter param = AudioParameter(keyValuePair); 5405 int value; 5406 audio_format_t reqFormat = mFormat; 5407 int reqSamplingRate = mReqSampleRate; 5408 int reqChannelCount = mReqChannelCount; 5409 5410 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5411 reqSamplingRate = value; 5412 reconfig = true; 5413 } 5414 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5415 reqFormat = (audio_format_t) value; 5416 reconfig = true; 5417 } 5418 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5419 reqChannelCount = popcount(value); 5420 reconfig = true; 5421 } 5422 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5423 // do not accept frame count changes if tracks are open as the track buffer 5424 // size depends on frame count and correct behavior would not be guaranteed 5425 // if frame count is changed after track creation 5426 if (mActiveTrack != 0) { 5427 status = INVALID_OPERATION; 5428 } else { 5429 reconfig = true; 5430 } 5431 } 5432 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5433 // forward device change to effects that have requested to be 5434 // aware of attached audio device. 5435 for (size_t i = 0; i < mEffectChains.size(); i++) { 5436 mEffectChains[i]->setDevice_l(value); 5437 } 5438 // store input device and output device but do not forward output device to audio HAL. 5439 // Note that status is ignored by the caller for output device 5440 // (see AudioFlinger::setParameters() 5441 if (value & AUDIO_DEVICE_OUT_ALL) { 5442 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5443 status = BAD_VALUE; 5444 } else { 5445 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5446 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5447 if (mTrack != NULL) { 5448 bool suspend = audio_is_bluetooth_sco_device( 5449 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5450 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5451 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5452 } 5453 } 5454 mDevice |= (uint32_t)value; 5455 } 5456 if (status == NO_ERROR) { 5457 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5458 if (status == INVALID_OPERATION) { 5459 mInput->stream->common.standby(&mInput->stream->common); 5460 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5461 keyValuePair.string()); 5462 } 5463 if (reconfig) { 5464 if (status == BAD_VALUE && 5465 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5466 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5467 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5468 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5469 (reqChannelCount <= FCC_2)) { 5470 status = NO_ERROR; 5471 } 5472 if (status == NO_ERROR) { 5473 readInputParameters(); 5474 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5475 } 5476 } 5477 } 5478 5479 mNewParameters.removeAt(0); 5480 5481 mParamStatus = status; 5482 mParamCond.signal(); 5483 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5484 // already timed out waiting for the status and will never signal the condition. 5485 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5486 } 5487 return reconfig; 5488} 5489 5490String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5491{ 5492 char *s; 5493 String8 out_s8 = String8(); 5494 5495 Mutex::Autolock _l(mLock); 5496 if (initCheck() != NO_ERROR) { 5497 return out_s8; 5498 } 5499 5500 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5501 out_s8 = String8(s); 5502 free(s); 5503 return out_s8; 5504} 5505 5506void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5507 AudioSystem::OutputDescriptor desc; 5508 void *param2 = NULL; 5509 5510 switch (event) { 5511 case AudioSystem::INPUT_OPENED: 5512 case AudioSystem::INPUT_CONFIG_CHANGED: 5513 desc.channels = mChannelMask; 5514 desc.samplingRate = mSampleRate; 5515 desc.format = mFormat; 5516 desc.frameCount = mFrameCount; 5517 desc.latency = 0; 5518 param2 = &desc; 5519 break; 5520 5521 case AudioSystem::INPUT_CLOSED: 5522 default: 5523 break; 5524 } 5525 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5526} 5527 5528void AudioFlinger::RecordThread::readInputParameters() 5529{ 5530 delete mRsmpInBuffer; 5531 // mRsmpInBuffer is always assigned a new[] below 5532 delete mRsmpOutBuffer; 5533 mRsmpOutBuffer = NULL; 5534 delete mResampler; 5535 mResampler = NULL; 5536 5537 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5538 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5539 mChannelCount = (uint16_t)popcount(mChannelMask); 5540 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5541 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5542 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5543 mFrameCount = mInputBytes / mFrameSize; 5544 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5545 5546 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5547 { 5548 int channelCount; 5549 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5550 // stereo to mono post process as the resampler always outputs stereo. 5551 if (mChannelCount == 1 && mReqChannelCount == 2) { 5552 channelCount = 1; 5553 } else { 5554 channelCount = 2; 5555 } 5556 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5557 mResampler->setSampleRate(mSampleRate); 5558 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5559 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5560 5561 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5562 if (mChannelCount == 1 && mReqChannelCount == 1) { 5563 mFrameCount >>= 1; 5564 } 5565 5566 } 5567 mRsmpInIndex = mFrameCount; 5568} 5569 5570unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5571{ 5572 Mutex::Autolock _l(mLock); 5573 if (initCheck() != NO_ERROR) { 5574 return 0; 5575 } 5576 5577 return mInput->stream->get_input_frames_lost(mInput->stream); 5578} 5579 5580uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5581{ 5582 Mutex::Autolock _l(mLock); 5583 uint32_t result = 0; 5584 if (getEffectChain_l(sessionId) != 0) { 5585 result = EFFECT_SESSION; 5586 } 5587 5588 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5589 result |= TRACK_SESSION; 5590 } 5591 5592 return result; 5593} 5594 5595AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5596{ 5597 Mutex::Autolock _l(mLock); 5598 return mTrack; 5599} 5600 5601AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5602{ 5603 Mutex::Autolock _l(mLock); 5604 return mInput; 5605} 5606 5607AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5608{ 5609 Mutex::Autolock _l(mLock); 5610 AudioStreamIn *input = mInput; 5611 mInput = NULL; 5612 return input; 5613} 5614 5615// this method must always be called either with ThreadBase mLock held or inside the thread loop 5616audio_stream_t* AudioFlinger::RecordThread::stream() 5617{ 5618 if (mInput == NULL) { 5619 return NULL; 5620 } 5621 return &mInput->stream->common; 5622} 5623 5624 5625// ---------------------------------------------------------------------------- 5626 5627audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5628 uint32_t *pSamplingRate, 5629 audio_format_t *pFormat, 5630 uint32_t *pChannels, 5631 uint32_t *pLatencyMs, 5632 audio_policy_output_flags_t flags) 5633{ 5634 status_t status; 5635 PlaybackThread *thread = NULL; 5636 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5637 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5638 uint32_t channels = pChannels ? *pChannels : 0; 5639 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5640 audio_stream_out_t *outStream; 5641 audio_hw_device_t *outHwDev; 5642 5643 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5644 pDevices ? *pDevices : 0, 5645 samplingRate, 5646 format, 5647 channels, 5648 flags); 5649 5650 if (pDevices == NULL || *pDevices == 0) { 5651 return 0; 5652 } 5653 5654 Mutex::Autolock _l(mLock); 5655 5656 outHwDev = findSuitableHwDev_l(*pDevices); 5657 if (outHwDev == NULL) 5658 return 0; 5659 5660 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5661 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5662 &channels, &samplingRate, &outStream); 5663 mHardwareStatus = AUDIO_HW_IDLE; 5664 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5665 outStream, 5666 samplingRate, 5667 format, 5668 channels, 5669 status); 5670 5671 if (outStream != NULL) { 5672 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5673 audio_io_handle_t id = nextUniqueId(); 5674 5675 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5676 (format != AUDIO_FORMAT_PCM_16_BIT) || 5677 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5678 thread = new DirectOutputThread(this, output, id, *pDevices); 5679 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5680 } else { 5681 thread = new MixerThread(this, output, id, *pDevices); 5682 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5683 } 5684 mPlaybackThreads.add(id, thread); 5685 5686 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5687 if (pFormat != NULL) *pFormat = format; 5688 if (pChannels != NULL) *pChannels = channels; 5689 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5690 5691 // notify client processes of the new output creation 5692 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5693 return id; 5694 } 5695 5696 return 0; 5697} 5698 5699audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5700 audio_io_handle_t output2) 5701{ 5702 Mutex::Autolock _l(mLock); 5703 MixerThread *thread1 = checkMixerThread_l(output1); 5704 MixerThread *thread2 = checkMixerThread_l(output2); 5705 5706 if (thread1 == NULL || thread2 == NULL) { 5707 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5708 return 0; 5709 } 5710 5711 audio_io_handle_t id = nextUniqueId(); 5712 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5713 thread->addOutputTrack(thread2); 5714 mPlaybackThreads.add(id, thread); 5715 // notify client processes of the new output creation 5716 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5717 return id; 5718} 5719 5720status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5721{ 5722 // keep strong reference on the playback thread so that 5723 // it is not destroyed while exit() is executed 5724 sp<PlaybackThread> thread; 5725 { 5726 Mutex::Autolock _l(mLock); 5727 thread = checkPlaybackThread_l(output); 5728 if (thread == NULL) { 5729 return BAD_VALUE; 5730 } 5731 5732 ALOGV("closeOutput() %d", output); 5733 5734 if (thread->type() == ThreadBase::MIXER) { 5735 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5736 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5737 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5738 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5739 } 5740 } 5741 } 5742 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5743 mPlaybackThreads.removeItem(output); 5744 } 5745 thread->exit(); 5746 // The thread entity (active unit of execution) is no longer running here, 5747 // but the ThreadBase container still exists. 5748 5749 if (thread->type() != ThreadBase::DUPLICATING) { 5750 AudioStreamOut *out = thread->clearOutput(); 5751 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5752 // from now on thread->mOutput is NULL 5753 out->hwDev->close_output_stream(out->hwDev, out->stream); 5754 delete out; 5755 } 5756 return NO_ERROR; 5757} 5758 5759status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5760{ 5761 Mutex::Autolock _l(mLock); 5762 PlaybackThread *thread = checkPlaybackThread_l(output); 5763 5764 if (thread == NULL) { 5765 return BAD_VALUE; 5766 } 5767 5768 ALOGV("suspendOutput() %d", output); 5769 thread->suspend(); 5770 5771 return NO_ERROR; 5772} 5773 5774status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5775{ 5776 Mutex::Autolock _l(mLock); 5777 PlaybackThread *thread = checkPlaybackThread_l(output); 5778 5779 if (thread == NULL) { 5780 return BAD_VALUE; 5781 } 5782 5783 ALOGV("restoreOutput() %d", output); 5784 5785 thread->restore(); 5786 5787 return NO_ERROR; 5788} 5789 5790audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5791 uint32_t *pSamplingRate, 5792 audio_format_t *pFormat, 5793 uint32_t *pChannels, 5794 audio_in_acoustics_t acoustics) 5795{ 5796 status_t status; 5797 RecordThread *thread = NULL; 5798 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5799 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5800 uint32_t channels = pChannels ? *pChannels : 0; 5801 uint32_t reqSamplingRate = samplingRate; 5802 audio_format_t reqFormat = format; 5803 uint32_t reqChannels = channels; 5804 audio_stream_in_t *inStream; 5805 audio_hw_device_t *inHwDev; 5806 5807 if (pDevices == NULL || *pDevices == 0) { 5808 return 0; 5809 } 5810 5811 Mutex::Autolock _l(mLock); 5812 5813 inHwDev = findSuitableHwDev_l(*pDevices); 5814 if (inHwDev == NULL) 5815 return 0; 5816 5817 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5818 &channels, &samplingRate, 5819 acoustics, 5820 &inStream); 5821 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5822 inStream, 5823 samplingRate, 5824 format, 5825 channels, 5826 acoustics, 5827 status); 5828 5829 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5830 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5831 // or stereo to mono conversions on 16 bit PCM inputs. 5832 if (inStream == NULL && status == BAD_VALUE && 5833 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5834 (samplingRate <= 2 * reqSamplingRate) && 5835 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5836 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5837 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5838 &channels, &samplingRate, 5839 acoustics, 5840 &inStream); 5841 } 5842 5843 if (inStream != NULL) { 5844 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5845 5846 audio_io_handle_t id = nextUniqueId(); 5847 // Start record thread 5848 // RecorThread require both input and output device indication to forward to audio 5849 // pre processing modules 5850 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5851 thread = new RecordThread(this, 5852 input, 5853 reqSamplingRate, 5854 reqChannels, 5855 id, 5856 device); 5857 mRecordThreads.add(id, thread); 5858 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5859 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5860 if (pFormat != NULL) *pFormat = format; 5861 if (pChannels != NULL) *pChannels = reqChannels; 5862 5863 input->stream->common.standby(&input->stream->common); 5864 5865 // notify client processes of the new input creation 5866 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5867 return id; 5868 } 5869 5870 return 0; 5871} 5872 5873status_t AudioFlinger::closeInput(audio_io_handle_t input) 5874{ 5875 // keep strong reference on the record thread so that 5876 // it is not destroyed while exit() is executed 5877 sp<RecordThread> thread; 5878 { 5879 Mutex::Autolock _l(mLock); 5880 thread = checkRecordThread_l(input); 5881 if (thread == NULL) { 5882 return BAD_VALUE; 5883 } 5884 5885 ALOGV("closeInput() %d", input); 5886 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5887 mRecordThreads.removeItem(input); 5888 } 5889 thread->exit(); 5890 // The thread entity (active unit of execution) is no longer running here, 5891 // but the ThreadBase container still exists. 5892 5893 AudioStreamIn *in = thread->clearInput(); 5894 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5895 // from now on thread->mInput is NULL 5896 in->hwDev->close_input_stream(in->hwDev, in->stream); 5897 delete in; 5898 5899 return NO_ERROR; 5900} 5901 5902status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5903{ 5904 Mutex::Autolock _l(mLock); 5905 MixerThread *dstThread = checkMixerThread_l(output); 5906 if (dstThread == NULL) { 5907 ALOGW("setStreamOutput() bad output id %d", output); 5908 return BAD_VALUE; 5909 } 5910 5911 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5912 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5913 5914 dstThread->setStreamValid(stream, true); 5915 5916 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5917 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5918 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5919 MixerThread *srcThread = (MixerThread *)thread; 5920 srcThread->setStreamValid(stream, false); 5921 srcThread->invalidateTracks(stream); 5922 } 5923 } 5924 5925 return NO_ERROR; 5926} 5927 5928 5929int AudioFlinger::newAudioSessionId() 5930{ 5931 return nextUniqueId(); 5932} 5933 5934void AudioFlinger::acquireAudioSessionId(int audioSession) 5935{ 5936 Mutex::Autolock _l(mLock); 5937 pid_t caller = IPCThreadState::self()->getCallingPid(); 5938 ALOGV("acquiring %d from %d", audioSession, caller); 5939 size_t num = mAudioSessionRefs.size(); 5940 for (size_t i = 0; i< num; i++) { 5941 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5942 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5943 ref->mCnt++; 5944 ALOGV(" incremented refcount to %d", ref->mCnt); 5945 return; 5946 } 5947 } 5948 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5949 ALOGV(" added new entry for %d", audioSession); 5950} 5951 5952void AudioFlinger::releaseAudioSessionId(int audioSession) 5953{ 5954 Mutex::Autolock _l(mLock); 5955 pid_t caller = IPCThreadState::self()->getCallingPid(); 5956 ALOGV("releasing %d from %d", audioSession, caller); 5957 size_t num = mAudioSessionRefs.size(); 5958 for (size_t i = 0; i< num; i++) { 5959 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5960 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5961 ref->mCnt--; 5962 ALOGV(" decremented refcount to %d", ref->mCnt); 5963 if (ref->mCnt == 0) { 5964 mAudioSessionRefs.removeAt(i); 5965 delete ref; 5966 purgeStaleEffects_l(); 5967 } 5968 return; 5969 } 5970 } 5971 ALOGW("session id %d not found for pid %d", audioSession, caller); 5972} 5973 5974void AudioFlinger::purgeStaleEffects_l() { 5975 5976 ALOGV("purging stale effects"); 5977 5978 Vector< sp<EffectChain> > chains; 5979 5980 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5981 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5982 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5983 sp<EffectChain> ec = t->mEffectChains[j]; 5984 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5985 chains.push(ec); 5986 } 5987 } 5988 } 5989 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5990 sp<RecordThread> t = mRecordThreads.valueAt(i); 5991 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5992 sp<EffectChain> ec = t->mEffectChains[j]; 5993 chains.push(ec); 5994 } 5995 } 5996 5997 for (size_t i = 0; i < chains.size(); i++) { 5998 sp<EffectChain> ec = chains[i]; 5999 int sessionid = ec->sessionId(); 6000 sp<ThreadBase> t = ec->mThread.promote(); 6001 if (t == 0) { 6002 continue; 6003 } 6004 size_t numsessionrefs = mAudioSessionRefs.size(); 6005 bool found = false; 6006 for (size_t k = 0; k < numsessionrefs; k++) { 6007 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6008 if (ref->mSessionid == sessionid) { 6009 ALOGV(" session %d still exists for %d with %d refs", 6010 sessionid, ref->mPid, ref->mCnt); 6011 found = true; 6012 break; 6013 } 6014 } 6015 if (!found) { 6016 // remove all effects from the chain 6017 while (ec->mEffects.size()) { 6018 sp<EffectModule> effect = ec->mEffects[0]; 6019 effect->unPin(); 6020 Mutex::Autolock _l (t->mLock); 6021 t->removeEffect_l(effect); 6022 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6023 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6024 if (handle != 0) { 6025 handle->mEffect.clear(); 6026 if (handle->mHasControl && handle->mEnabled) { 6027 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6028 } 6029 } 6030 } 6031 AudioSystem::unregisterEffect(effect->id()); 6032 } 6033 } 6034 } 6035 return; 6036} 6037 6038// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6039AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6040{ 6041 return mPlaybackThreads.valueFor(output).get(); 6042} 6043 6044// checkMixerThread_l() must be called with AudioFlinger::mLock held 6045AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6046{ 6047 PlaybackThread *thread = checkPlaybackThread_l(output); 6048 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6049} 6050 6051// checkRecordThread_l() must be called with AudioFlinger::mLock held 6052AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6053{ 6054 return mRecordThreads.valueFor(input).get(); 6055} 6056 6057uint32_t AudioFlinger::nextUniqueId() 6058{ 6059 return android_atomic_inc(&mNextUniqueId); 6060} 6061 6062AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6063{ 6064 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6065 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6066 AudioStreamOut *output = thread->getOutput(); 6067 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6068 return thread; 6069 } 6070 } 6071 return NULL; 6072} 6073 6074uint32_t AudioFlinger::primaryOutputDevice_l() const 6075{ 6076 PlaybackThread *thread = primaryPlaybackThread_l(); 6077 6078 if (thread == NULL) { 6079 return 0; 6080 } 6081 6082 return thread->device(); 6083} 6084 6085sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6086 int triggerSession, 6087 int listenerSession, 6088 sync_event_callback_t callBack, 6089 void *cookie) 6090{ 6091 Mutex::Autolock _l(mLock); 6092 6093 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6094 status_t playStatus = NAME_NOT_FOUND; 6095 status_t recStatus = NAME_NOT_FOUND; 6096 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6097 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6098 if (playStatus == NO_ERROR) { 6099 return event; 6100 } 6101 } 6102 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6103 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6104 if (recStatus == NO_ERROR) { 6105 return event; 6106 } 6107 } 6108 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6109 mPendingSyncEvents.add(event); 6110 } else { 6111 ALOGV("createSyncEvent() invalid event %d", event->type()); 6112 event.clear(); 6113 } 6114 return event; 6115} 6116 6117// ---------------------------------------------------------------------------- 6118// Effect management 6119// ---------------------------------------------------------------------------- 6120 6121 6122status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6123{ 6124 Mutex::Autolock _l(mLock); 6125 return EffectQueryNumberEffects(numEffects); 6126} 6127 6128status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6129{ 6130 Mutex::Autolock _l(mLock); 6131 return EffectQueryEffect(index, descriptor); 6132} 6133 6134status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6135 effect_descriptor_t *descriptor) const 6136{ 6137 Mutex::Autolock _l(mLock); 6138 return EffectGetDescriptor(pUuid, descriptor); 6139} 6140 6141 6142sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6143 effect_descriptor_t *pDesc, 6144 const sp<IEffectClient>& effectClient, 6145 int32_t priority, 6146 audio_io_handle_t io, 6147 int sessionId, 6148 status_t *status, 6149 int *id, 6150 int *enabled) 6151{ 6152 status_t lStatus = NO_ERROR; 6153 sp<EffectHandle> handle; 6154 effect_descriptor_t desc; 6155 6156 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6157 pid, effectClient.get(), priority, sessionId, io); 6158 6159 if (pDesc == NULL) { 6160 lStatus = BAD_VALUE; 6161 goto Exit; 6162 } 6163 6164 // check audio settings permission for global effects 6165 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6166 lStatus = PERMISSION_DENIED; 6167 goto Exit; 6168 } 6169 6170 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6171 // that can only be created by audio policy manager (running in same process) 6172 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6173 lStatus = PERMISSION_DENIED; 6174 goto Exit; 6175 } 6176 6177 if (io == 0) { 6178 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6179 // output must be specified by AudioPolicyManager when using session 6180 // AUDIO_SESSION_OUTPUT_STAGE 6181 lStatus = BAD_VALUE; 6182 goto Exit; 6183 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6184 // if the output returned by getOutputForEffect() is removed before we lock the 6185 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6186 // and we will exit safely 6187 io = AudioSystem::getOutputForEffect(&desc); 6188 } 6189 } 6190 6191 { 6192 Mutex::Autolock _l(mLock); 6193 6194 6195 if (!EffectIsNullUuid(&pDesc->uuid)) { 6196 // if uuid is specified, request effect descriptor 6197 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6198 if (lStatus < 0) { 6199 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6200 goto Exit; 6201 } 6202 } else { 6203 // if uuid is not specified, look for an available implementation 6204 // of the required type in effect factory 6205 if (EffectIsNullUuid(&pDesc->type)) { 6206 ALOGW("createEffect() no effect type"); 6207 lStatus = BAD_VALUE; 6208 goto Exit; 6209 } 6210 uint32_t numEffects = 0; 6211 effect_descriptor_t d; 6212 d.flags = 0; // prevent compiler warning 6213 bool found = false; 6214 6215 lStatus = EffectQueryNumberEffects(&numEffects); 6216 if (lStatus < 0) { 6217 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6218 goto Exit; 6219 } 6220 for (uint32_t i = 0; i < numEffects; i++) { 6221 lStatus = EffectQueryEffect(i, &desc); 6222 if (lStatus < 0) { 6223 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6224 continue; 6225 } 6226 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6227 // If matching type found save effect descriptor. If the session is 6228 // 0 and the effect is not auxiliary, continue enumeration in case 6229 // an auxiliary version of this effect type is available 6230 found = true; 6231 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6232 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6233 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6234 break; 6235 } 6236 } 6237 } 6238 if (!found) { 6239 lStatus = BAD_VALUE; 6240 ALOGW("createEffect() effect not found"); 6241 goto Exit; 6242 } 6243 // For same effect type, chose auxiliary version over insert version if 6244 // connect to output mix (Compliance to OpenSL ES) 6245 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6246 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6247 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6248 } 6249 } 6250 6251 // Do not allow auxiliary effects on a session different from 0 (output mix) 6252 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6253 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6254 lStatus = INVALID_OPERATION; 6255 goto Exit; 6256 } 6257 6258 // check recording permission for visualizer 6259 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6260 !recordingAllowed()) { 6261 lStatus = PERMISSION_DENIED; 6262 goto Exit; 6263 } 6264 6265 // return effect descriptor 6266 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6267 6268 // If output is not specified try to find a matching audio session ID in one of the 6269 // output threads. 6270 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6271 // because of code checking output when entering the function. 6272 // Note: io is never 0 when creating an effect on an input 6273 if (io == 0) { 6274 // look for the thread where the specified audio session is present 6275 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6276 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6277 io = mPlaybackThreads.keyAt(i); 6278 break; 6279 } 6280 } 6281 if (io == 0) { 6282 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6283 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6284 io = mRecordThreads.keyAt(i); 6285 break; 6286 } 6287 } 6288 } 6289 // If no output thread contains the requested session ID, default to 6290 // first output. The effect chain will be moved to the correct output 6291 // thread when a track with the same session ID is created 6292 if (io == 0 && mPlaybackThreads.size()) { 6293 io = mPlaybackThreads.keyAt(0); 6294 } 6295 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6296 } 6297 ThreadBase *thread = checkRecordThread_l(io); 6298 if (thread == NULL) { 6299 thread = checkPlaybackThread_l(io); 6300 if (thread == NULL) { 6301 ALOGE("createEffect() unknown output thread"); 6302 lStatus = BAD_VALUE; 6303 goto Exit; 6304 } 6305 } 6306 6307 sp<Client> client = registerPid_l(pid); 6308 6309 // create effect on selected output thread 6310 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6311 &desc, enabled, &lStatus); 6312 if (handle != 0 && id != NULL) { 6313 *id = handle->id(); 6314 } 6315 } 6316 6317Exit: 6318 if (status != NULL) { 6319 *status = lStatus; 6320 } 6321 return handle; 6322} 6323 6324status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6325 audio_io_handle_t dstOutput) 6326{ 6327 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6328 sessionId, srcOutput, dstOutput); 6329 Mutex::Autolock _l(mLock); 6330 if (srcOutput == dstOutput) { 6331 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6332 return NO_ERROR; 6333 } 6334 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6335 if (srcThread == NULL) { 6336 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6337 return BAD_VALUE; 6338 } 6339 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6340 if (dstThread == NULL) { 6341 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6342 return BAD_VALUE; 6343 } 6344 6345 Mutex::Autolock _dl(dstThread->mLock); 6346 Mutex::Autolock _sl(srcThread->mLock); 6347 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6348 6349 return NO_ERROR; 6350} 6351 6352// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6353status_t AudioFlinger::moveEffectChain_l(int sessionId, 6354 AudioFlinger::PlaybackThread *srcThread, 6355 AudioFlinger::PlaybackThread *dstThread, 6356 bool reRegister) 6357{ 6358 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6359 sessionId, srcThread, dstThread); 6360 6361 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6362 if (chain == 0) { 6363 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6364 sessionId, srcThread); 6365 return INVALID_OPERATION; 6366 } 6367 6368 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6369 // so that a new chain is created with correct parameters when first effect is added. This is 6370 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6371 // removed. 6372 srcThread->removeEffectChain_l(chain); 6373 6374 // transfer all effects one by one so that new effect chain is created on new thread with 6375 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6376 audio_io_handle_t dstOutput = dstThread->id(); 6377 sp<EffectChain> dstChain; 6378 uint32_t strategy = 0; // prevent compiler warning 6379 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6380 while (effect != 0) { 6381 srcThread->removeEffect_l(effect); 6382 dstThread->addEffect_l(effect); 6383 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6384 if (effect->state() == EffectModule::ACTIVE || 6385 effect->state() == EffectModule::STOPPING) { 6386 effect->start(); 6387 } 6388 // if the move request is not received from audio policy manager, the effect must be 6389 // re-registered with the new strategy and output 6390 if (dstChain == 0) { 6391 dstChain = effect->chain().promote(); 6392 if (dstChain == 0) { 6393 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6394 srcThread->addEffect_l(effect); 6395 return NO_INIT; 6396 } 6397 strategy = dstChain->strategy(); 6398 } 6399 if (reRegister) { 6400 AudioSystem::unregisterEffect(effect->id()); 6401 AudioSystem::registerEffect(&effect->desc(), 6402 dstOutput, 6403 strategy, 6404 sessionId, 6405 effect->id()); 6406 } 6407 effect = chain->getEffectFromId_l(0); 6408 } 6409 6410 return NO_ERROR; 6411} 6412 6413 6414// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6415sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6416 const sp<AudioFlinger::Client>& client, 6417 const sp<IEffectClient>& effectClient, 6418 int32_t priority, 6419 int sessionId, 6420 effect_descriptor_t *desc, 6421 int *enabled, 6422 status_t *status 6423 ) 6424{ 6425 sp<EffectModule> effect; 6426 sp<EffectHandle> handle; 6427 status_t lStatus; 6428 sp<EffectChain> chain; 6429 bool chainCreated = false; 6430 bool effectCreated = false; 6431 bool effectRegistered = false; 6432 6433 lStatus = initCheck(); 6434 if (lStatus != NO_ERROR) { 6435 ALOGW("createEffect_l() Audio driver not initialized."); 6436 goto Exit; 6437 } 6438 6439 // Do not allow effects with session ID 0 on direct output or duplicating threads 6440 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6441 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6442 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6443 desc->name, sessionId); 6444 lStatus = BAD_VALUE; 6445 goto Exit; 6446 } 6447 // Only Pre processor effects are allowed on input threads and only on input threads 6448 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6449 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6450 desc->name, desc->flags, mType); 6451 lStatus = BAD_VALUE; 6452 goto Exit; 6453 } 6454 6455 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6456 6457 { // scope for mLock 6458 Mutex::Autolock _l(mLock); 6459 6460 // check for existing effect chain with the requested audio session 6461 chain = getEffectChain_l(sessionId); 6462 if (chain == 0) { 6463 // create a new chain for this session 6464 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6465 chain = new EffectChain(this, sessionId); 6466 addEffectChain_l(chain); 6467 chain->setStrategy(getStrategyForSession_l(sessionId)); 6468 chainCreated = true; 6469 } else { 6470 effect = chain->getEffectFromDesc_l(desc); 6471 } 6472 6473 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6474 6475 if (effect == 0) { 6476 int id = mAudioFlinger->nextUniqueId(); 6477 // Check CPU and memory usage 6478 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6479 if (lStatus != NO_ERROR) { 6480 goto Exit; 6481 } 6482 effectRegistered = true; 6483 // create a new effect module if none present in the chain 6484 effect = new EffectModule(this, chain, desc, id, sessionId); 6485 lStatus = effect->status(); 6486 if (lStatus != NO_ERROR) { 6487 goto Exit; 6488 } 6489 lStatus = chain->addEffect_l(effect); 6490 if (lStatus != NO_ERROR) { 6491 goto Exit; 6492 } 6493 effectCreated = true; 6494 6495 effect->setDevice(mDevice); 6496 effect->setMode(mAudioFlinger->getMode()); 6497 } 6498 // create effect handle and connect it to effect module 6499 handle = new EffectHandle(effect, client, effectClient, priority); 6500 lStatus = effect->addHandle(handle); 6501 if (enabled != NULL) { 6502 *enabled = (int)effect->isEnabled(); 6503 } 6504 } 6505 6506Exit: 6507 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6508 Mutex::Autolock _l(mLock); 6509 if (effectCreated) { 6510 chain->removeEffect_l(effect); 6511 } 6512 if (effectRegistered) { 6513 AudioSystem::unregisterEffect(effect->id()); 6514 } 6515 if (chainCreated) { 6516 removeEffectChain_l(chain); 6517 } 6518 handle.clear(); 6519 } 6520 6521 if (status != NULL) { 6522 *status = lStatus; 6523 } 6524 return handle; 6525} 6526 6527sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6528{ 6529 sp<EffectChain> chain = getEffectChain_l(sessionId); 6530 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6531} 6532 6533// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6534// PlaybackThread::mLock held 6535status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6536{ 6537 // check for existing effect chain with the requested audio session 6538 int sessionId = effect->sessionId(); 6539 sp<EffectChain> chain = getEffectChain_l(sessionId); 6540 bool chainCreated = false; 6541 6542 if (chain == 0) { 6543 // create a new chain for this session 6544 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6545 chain = new EffectChain(this, sessionId); 6546 addEffectChain_l(chain); 6547 chain->setStrategy(getStrategyForSession_l(sessionId)); 6548 chainCreated = true; 6549 } 6550 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6551 6552 if (chain->getEffectFromId_l(effect->id()) != 0) { 6553 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6554 this, effect->desc().name, chain.get()); 6555 return BAD_VALUE; 6556 } 6557 6558 status_t status = chain->addEffect_l(effect); 6559 if (status != NO_ERROR) { 6560 if (chainCreated) { 6561 removeEffectChain_l(chain); 6562 } 6563 return status; 6564 } 6565 6566 effect->setDevice(mDevice); 6567 effect->setMode(mAudioFlinger->getMode()); 6568 return NO_ERROR; 6569} 6570 6571void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6572 6573 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6574 effect_descriptor_t desc = effect->desc(); 6575 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6576 detachAuxEffect_l(effect->id()); 6577 } 6578 6579 sp<EffectChain> chain = effect->chain().promote(); 6580 if (chain != 0) { 6581 // remove effect chain if removing last effect 6582 if (chain->removeEffect_l(effect) == 0) { 6583 removeEffectChain_l(chain); 6584 } 6585 } else { 6586 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6587 } 6588} 6589 6590void AudioFlinger::ThreadBase::lockEffectChains_l( 6591 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6592{ 6593 effectChains = mEffectChains; 6594 for (size_t i = 0; i < mEffectChains.size(); i++) { 6595 mEffectChains[i]->lock(); 6596 } 6597} 6598 6599void AudioFlinger::ThreadBase::unlockEffectChains( 6600 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6601{ 6602 for (size_t i = 0; i < effectChains.size(); i++) { 6603 effectChains[i]->unlock(); 6604 } 6605} 6606 6607sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6608{ 6609 Mutex::Autolock _l(mLock); 6610 return getEffectChain_l(sessionId); 6611} 6612 6613sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6614{ 6615 size_t size = mEffectChains.size(); 6616 for (size_t i = 0; i < size; i++) { 6617 if (mEffectChains[i]->sessionId() == sessionId) { 6618 return mEffectChains[i]; 6619 } 6620 } 6621 return 0; 6622} 6623 6624void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6625{ 6626 Mutex::Autolock _l(mLock); 6627 size_t size = mEffectChains.size(); 6628 for (size_t i = 0; i < size; i++) { 6629 mEffectChains[i]->setMode_l(mode); 6630 } 6631} 6632 6633void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6634 const wp<EffectHandle>& handle, 6635 bool unpinIfLast) { 6636 6637 Mutex::Autolock _l(mLock); 6638 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6639 // delete the effect module if removing last handle on it 6640 if (effect->removeHandle(handle) == 0) { 6641 if (!effect->isPinned() || unpinIfLast) { 6642 removeEffect_l(effect); 6643 AudioSystem::unregisterEffect(effect->id()); 6644 } 6645 } 6646} 6647 6648status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6649{ 6650 int session = chain->sessionId(); 6651 int16_t *buffer = mMixBuffer; 6652 bool ownsBuffer = false; 6653 6654 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6655 if (session > 0) { 6656 // Only one effect chain can be present in direct output thread and it uses 6657 // the mix buffer as input 6658 if (mType != DIRECT) { 6659 size_t numSamples = mFrameCount * mChannelCount; 6660 buffer = new int16_t[numSamples]; 6661 memset(buffer, 0, numSamples * sizeof(int16_t)); 6662 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6663 ownsBuffer = true; 6664 } 6665 6666 // Attach all tracks with same session ID to this chain. 6667 for (size_t i = 0; i < mTracks.size(); ++i) { 6668 sp<Track> track = mTracks[i]; 6669 if (session == track->sessionId()) { 6670 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6671 track->setMainBuffer(buffer); 6672 chain->incTrackCnt(); 6673 } 6674 } 6675 6676 // indicate all active tracks in the chain 6677 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6678 sp<Track> track = mActiveTracks[i].promote(); 6679 if (track == 0) continue; 6680 if (session == track->sessionId()) { 6681 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6682 chain->incActiveTrackCnt(); 6683 } 6684 } 6685 } 6686 6687 chain->setInBuffer(buffer, ownsBuffer); 6688 chain->setOutBuffer(mMixBuffer); 6689 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6690 // chains list in order to be processed last as it contains output stage effects 6691 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6692 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6693 // after track specific effects and before output stage 6694 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6695 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6696 // Effect chain for other sessions are inserted at beginning of effect 6697 // chains list to be processed before output mix effects. Relative order between other 6698 // sessions is not important 6699 size_t size = mEffectChains.size(); 6700 size_t i = 0; 6701 for (i = 0; i < size; i++) { 6702 if (mEffectChains[i]->sessionId() < session) break; 6703 } 6704 mEffectChains.insertAt(chain, i); 6705 checkSuspendOnAddEffectChain_l(chain); 6706 6707 return NO_ERROR; 6708} 6709 6710size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6711{ 6712 int session = chain->sessionId(); 6713 6714 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6715 6716 for (size_t i = 0; i < mEffectChains.size(); i++) { 6717 if (chain == mEffectChains[i]) { 6718 mEffectChains.removeAt(i); 6719 // detach all active tracks from the chain 6720 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6721 sp<Track> track = mActiveTracks[i].promote(); 6722 if (track == 0) continue; 6723 if (session == track->sessionId()) { 6724 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6725 chain.get(), session); 6726 chain->decActiveTrackCnt(); 6727 } 6728 } 6729 6730 // detach all tracks with same session ID from this chain 6731 for (size_t i = 0; i < mTracks.size(); ++i) { 6732 sp<Track> track = mTracks[i]; 6733 if (session == track->sessionId()) { 6734 track->setMainBuffer(mMixBuffer); 6735 chain->decTrackCnt(); 6736 } 6737 } 6738 break; 6739 } 6740 } 6741 return mEffectChains.size(); 6742} 6743 6744status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6745 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6746{ 6747 Mutex::Autolock _l(mLock); 6748 return attachAuxEffect_l(track, EffectId); 6749} 6750 6751status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6752 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6753{ 6754 status_t status = NO_ERROR; 6755 6756 if (EffectId == 0) { 6757 track->setAuxBuffer(0, NULL); 6758 } else { 6759 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6760 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6761 if (effect != 0) { 6762 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6763 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6764 } else { 6765 status = INVALID_OPERATION; 6766 } 6767 } else { 6768 status = BAD_VALUE; 6769 } 6770 } 6771 return status; 6772} 6773 6774void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6775{ 6776 for (size_t i = 0; i < mTracks.size(); ++i) { 6777 sp<Track> track = mTracks[i]; 6778 if (track->auxEffectId() == effectId) { 6779 attachAuxEffect_l(track, 0); 6780 } 6781 } 6782} 6783 6784status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6785{ 6786 // only one chain per input thread 6787 if (mEffectChains.size() != 0) { 6788 return INVALID_OPERATION; 6789 } 6790 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6791 6792 chain->setInBuffer(NULL); 6793 chain->setOutBuffer(NULL); 6794 6795 checkSuspendOnAddEffectChain_l(chain); 6796 6797 mEffectChains.add(chain); 6798 6799 return NO_ERROR; 6800} 6801 6802size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6803{ 6804 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6805 ALOGW_IF(mEffectChains.size() != 1, 6806 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6807 chain.get(), mEffectChains.size(), this); 6808 if (mEffectChains.size() == 1) { 6809 mEffectChains.removeAt(0); 6810 } 6811 return 0; 6812} 6813 6814// ---------------------------------------------------------------------------- 6815// EffectModule implementation 6816// ---------------------------------------------------------------------------- 6817 6818#undef LOG_TAG 6819#define LOG_TAG "AudioFlinger::EffectModule" 6820 6821AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6822 const wp<AudioFlinger::EffectChain>& chain, 6823 effect_descriptor_t *desc, 6824 int id, 6825 int sessionId) 6826 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6827 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6828{ 6829 ALOGV("Constructor %p", this); 6830 int lStatus; 6831 if (thread == NULL) { 6832 return; 6833 } 6834 6835 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6836 6837 // create effect engine from effect factory 6838 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6839 6840 if (mStatus != NO_ERROR) { 6841 return; 6842 } 6843 lStatus = init(); 6844 if (lStatus < 0) { 6845 mStatus = lStatus; 6846 goto Error; 6847 } 6848 6849 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6850 mPinned = true; 6851 } 6852 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6853 return; 6854Error: 6855 EffectRelease(mEffectInterface); 6856 mEffectInterface = NULL; 6857 ALOGV("Constructor Error %d", mStatus); 6858} 6859 6860AudioFlinger::EffectModule::~EffectModule() 6861{ 6862 ALOGV("Destructor %p", this); 6863 if (mEffectInterface != NULL) { 6864 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6865 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6866 sp<ThreadBase> thread = mThread.promote(); 6867 if (thread != 0) { 6868 audio_stream_t *stream = thread->stream(); 6869 if (stream != NULL) { 6870 stream->remove_audio_effect(stream, mEffectInterface); 6871 } 6872 } 6873 } 6874 // release effect engine 6875 EffectRelease(mEffectInterface); 6876 } 6877} 6878 6879status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6880{ 6881 status_t status; 6882 6883 Mutex::Autolock _l(mLock); 6884 int priority = handle->priority(); 6885 size_t size = mHandles.size(); 6886 sp<EffectHandle> h; 6887 size_t i; 6888 for (i = 0; i < size; i++) { 6889 h = mHandles[i].promote(); 6890 if (h == 0) continue; 6891 if (h->priority() <= priority) break; 6892 } 6893 // if inserted in first place, move effect control from previous owner to this handle 6894 if (i == 0) { 6895 bool enabled = false; 6896 if (h != 0) { 6897 enabled = h->enabled(); 6898 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6899 } 6900 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6901 status = NO_ERROR; 6902 } else { 6903 status = ALREADY_EXISTS; 6904 } 6905 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6906 mHandles.insertAt(handle, i); 6907 return status; 6908} 6909 6910size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6911{ 6912 Mutex::Autolock _l(mLock); 6913 size_t size = mHandles.size(); 6914 size_t i; 6915 for (i = 0; i < size; i++) { 6916 if (mHandles[i] == handle) break; 6917 } 6918 if (i == size) { 6919 return size; 6920 } 6921 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6922 6923 bool enabled = false; 6924 EffectHandle *hdl = handle.unsafe_get(); 6925 if (hdl != NULL) { 6926 ALOGV("removeHandle() unsafe_get OK"); 6927 enabled = hdl->enabled(); 6928 } 6929 mHandles.removeAt(i); 6930 size = mHandles.size(); 6931 // if removed from first place, move effect control from this handle to next in line 6932 if (i == 0 && size != 0) { 6933 sp<EffectHandle> h = mHandles[0].promote(); 6934 if (h != 0) { 6935 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6936 } 6937 } 6938 6939 // Prevent calls to process() and other functions on effect interface from now on. 6940 // The effect engine will be released by the destructor when the last strong reference on 6941 // this object is released which can happen after next process is called. 6942 if (size == 0 && !mPinned) { 6943 mState = DESTROYED; 6944 } 6945 6946 return size; 6947} 6948 6949sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6950{ 6951 Mutex::Autolock _l(mLock); 6952 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6953} 6954 6955void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6956{ 6957 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6958 // keep a strong reference on this EffectModule to avoid calling the 6959 // destructor before we exit 6960 sp<EffectModule> keep(this); 6961 { 6962 sp<ThreadBase> thread = mThread.promote(); 6963 if (thread != 0) { 6964 thread->disconnectEffect(keep, handle, unpinIfLast); 6965 } 6966 } 6967} 6968 6969void AudioFlinger::EffectModule::updateState() { 6970 Mutex::Autolock _l(mLock); 6971 6972 switch (mState) { 6973 case RESTART: 6974 reset_l(); 6975 // FALL THROUGH 6976 6977 case STARTING: 6978 // clear auxiliary effect input buffer for next accumulation 6979 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6980 memset(mConfig.inputCfg.buffer.raw, 6981 0, 6982 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6983 } 6984 start_l(); 6985 mState = ACTIVE; 6986 break; 6987 case STOPPING: 6988 stop_l(); 6989 mDisableWaitCnt = mMaxDisableWaitCnt; 6990 mState = STOPPED; 6991 break; 6992 case STOPPED: 6993 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6994 // turn off sequence. 6995 if (--mDisableWaitCnt == 0) { 6996 reset_l(); 6997 mState = IDLE; 6998 } 6999 break; 7000 default: //IDLE , ACTIVE, DESTROYED 7001 break; 7002 } 7003} 7004 7005void AudioFlinger::EffectModule::process() 7006{ 7007 Mutex::Autolock _l(mLock); 7008 7009 if (mState == DESTROYED || mEffectInterface == NULL || 7010 mConfig.inputCfg.buffer.raw == NULL || 7011 mConfig.outputCfg.buffer.raw == NULL) { 7012 return; 7013 } 7014 7015 if (isProcessEnabled()) { 7016 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7017 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7018 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7019 mConfig.inputCfg.buffer.s32, 7020 mConfig.inputCfg.buffer.frameCount/2); 7021 } 7022 7023 // do the actual processing in the effect engine 7024 int ret = (*mEffectInterface)->process(mEffectInterface, 7025 &mConfig.inputCfg.buffer, 7026 &mConfig.outputCfg.buffer); 7027 7028 // force transition to IDLE state when engine is ready 7029 if (mState == STOPPED && ret == -ENODATA) { 7030 mDisableWaitCnt = 1; 7031 } 7032 7033 // clear auxiliary effect input buffer for next accumulation 7034 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7035 memset(mConfig.inputCfg.buffer.raw, 0, 7036 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7037 } 7038 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7039 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7040 // If an insert effect is idle and input buffer is different from output buffer, 7041 // accumulate input onto output 7042 sp<EffectChain> chain = mChain.promote(); 7043 if (chain != 0 && chain->activeTrackCnt() != 0) { 7044 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7045 int16_t *in = mConfig.inputCfg.buffer.s16; 7046 int16_t *out = mConfig.outputCfg.buffer.s16; 7047 for (size_t i = 0; i < frameCnt; i++) { 7048 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7049 } 7050 } 7051 } 7052} 7053 7054void AudioFlinger::EffectModule::reset_l() 7055{ 7056 if (mEffectInterface == NULL) { 7057 return; 7058 } 7059 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7060} 7061 7062status_t AudioFlinger::EffectModule::configure() 7063{ 7064 uint32_t channels; 7065 if (mEffectInterface == NULL) { 7066 return NO_INIT; 7067 } 7068 7069 sp<ThreadBase> thread = mThread.promote(); 7070 if (thread == 0) { 7071 return DEAD_OBJECT; 7072 } 7073 7074 // TODO: handle configuration of effects replacing track process 7075 if (thread->channelCount() == 1) { 7076 channels = AUDIO_CHANNEL_OUT_MONO; 7077 } else { 7078 channels = AUDIO_CHANNEL_OUT_STEREO; 7079 } 7080 7081 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7082 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7083 } else { 7084 mConfig.inputCfg.channels = channels; 7085 } 7086 mConfig.outputCfg.channels = channels; 7087 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7088 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7089 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7090 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7091 mConfig.inputCfg.bufferProvider.cookie = NULL; 7092 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7093 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7094 mConfig.outputCfg.bufferProvider.cookie = NULL; 7095 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7096 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7097 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7098 // Insert effect: 7099 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7100 // always overwrites output buffer: input buffer == output buffer 7101 // - in other sessions: 7102 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7103 // other effect: overwrites output buffer: input buffer == output buffer 7104 // Auxiliary effect: 7105 // accumulates in output buffer: input buffer != output buffer 7106 // Therefore: accumulate <=> input buffer != output buffer 7107 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7108 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7109 } else { 7110 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7111 } 7112 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7113 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7114 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7115 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7116 7117 ALOGV("configure() %p thread %p buffer %p framecount %d", 7118 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7119 7120 status_t cmdStatus; 7121 uint32_t size = sizeof(int); 7122 status_t status = (*mEffectInterface)->command(mEffectInterface, 7123 EFFECT_CMD_SET_CONFIG, 7124 sizeof(effect_config_t), 7125 &mConfig, 7126 &size, 7127 &cmdStatus); 7128 if (status == 0) { 7129 status = cmdStatus; 7130 } 7131 7132 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7133 (1000 * mConfig.outputCfg.buffer.frameCount); 7134 7135 return status; 7136} 7137 7138status_t AudioFlinger::EffectModule::init() 7139{ 7140 Mutex::Autolock _l(mLock); 7141 if (mEffectInterface == NULL) { 7142 return NO_INIT; 7143 } 7144 status_t cmdStatus; 7145 uint32_t size = sizeof(status_t); 7146 status_t status = (*mEffectInterface)->command(mEffectInterface, 7147 EFFECT_CMD_INIT, 7148 0, 7149 NULL, 7150 &size, 7151 &cmdStatus); 7152 if (status == 0) { 7153 status = cmdStatus; 7154 } 7155 return status; 7156} 7157 7158status_t AudioFlinger::EffectModule::start() 7159{ 7160 Mutex::Autolock _l(mLock); 7161 return start_l(); 7162} 7163 7164status_t AudioFlinger::EffectModule::start_l() 7165{ 7166 if (mEffectInterface == NULL) { 7167 return NO_INIT; 7168 } 7169 status_t cmdStatus; 7170 uint32_t size = sizeof(status_t); 7171 status_t status = (*mEffectInterface)->command(mEffectInterface, 7172 EFFECT_CMD_ENABLE, 7173 0, 7174 NULL, 7175 &size, 7176 &cmdStatus); 7177 if (status == 0) { 7178 status = cmdStatus; 7179 } 7180 if (status == 0 && 7181 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7182 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7183 sp<ThreadBase> thread = mThread.promote(); 7184 if (thread != 0) { 7185 audio_stream_t *stream = thread->stream(); 7186 if (stream != NULL) { 7187 stream->add_audio_effect(stream, mEffectInterface); 7188 } 7189 } 7190 } 7191 return status; 7192} 7193 7194status_t AudioFlinger::EffectModule::stop() 7195{ 7196 Mutex::Autolock _l(mLock); 7197 return stop_l(); 7198} 7199 7200status_t AudioFlinger::EffectModule::stop_l() 7201{ 7202 if (mEffectInterface == NULL) { 7203 return NO_INIT; 7204 } 7205 status_t cmdStatus; 7206 uint32_t size = sizeof(status_t); 7207 status_t status = (*mEffectInterface)->command(mEffectInterface, 7208 EFFECT_CMD_DISABLE, 7209 0, 7210 NULL, 7211 &size, 7212 &cmdStatus); 7213 if (status == 0) { 7214 status = cmdStatus; 7215 } 7216 if (status == 0 && 7217 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7218 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7219 sp<ThreadBase> thread = mThread.promote(); 7220 if (thread != 0) { 7221 audio_stream_t *stream = thread->stream(); 7222 if (stream != NULL) { 7223 stream->remove_audio_effect(stream, mEffectInterface); 7224 } 7225 } 7226 } 7227 return status; 7228} 7229 7230status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7231 uint32_t cmdSize, 7232 void *pCmdData, 7233 uint32_t *replySize, 7234 void *pReplyData) 7235{ 7236 Mutex::Autolock _l(mLock); 7237// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7238 7239 if (mState == DESTROYED || mEffectInterface == NULL) { 7240 return NO_INIT; 7241 } 7242 status_t status = (*mEffectInterface)->command(mEffectInterface, 7243 cmdCode, 7244 cmdSize, 7245 pCmdData, 7246 replySize, 7247 pReplyData); 7248 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7249 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7250 for (size_t i = 1; i < mHandles.size(); i++) { 7251 sp<EffectHandle> h = mHandles[i].promote(); 7252 if (h != 0) { 7253 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7254 } 7255 } 7256 } 7257 return status; 7258} 7259 7260status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7261{ 7262 7263 Mutex::Autolock _l(mLock); 7264 ALOGV("setEnabled %p enabled %d", this, enabled); 7265 7266 if (enabled != isEnabled()) { 7267 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7268 if (enabled && status != NO_ERROR) { 7269 return status; 7270 } 7271 7272 switch (mState) { 7273 // going from disabled to enabled 7274 case IDLE: 7275 mState = STARTING; 7276 break; 7277 case STOPPED: 7278 mState = RESTART; 7279 break; 7280 case STOPPING: 7281 mState = ACTIVE; 7282 break; 7283 7284 // going from enabled to disabled 7285 case RESTART: 7286 mState = STOPPED; 7287 break; 7288 case STARTING: 7289 mState = IDLE; 7290 break; 7291 case ACTIVE: 7292 mState = STOPPING; 7293 break; 7294 case DESTROYED: 7295 return NO_ERROR; // simply ignore as we are being destroyed 7296 } 7297 for (size_t i = 1; i < mHandles.size(); i++) { 7298 sp<EffectHandle> h = mHandles[i].promote(); 7299 if (h != 0) { 7300 h->setEnabled(enabled); 7301 } 7302 } 7303 } 7304 return NO_ERROR; 7305} 7306 7307bool AudioFlinger::EffectModule::isEnabled() const 7308{ 7309 switch (mState) { 7310 case RESTART: 7311 case STARTING: 7312 case ACTIVE: 7313 return true; 7314 case IDLE: 7315 case STOPPING: 7316 case STOPPED: 7317 case DESTROYED: 7318 default: 7319 return false; 7320 } 7321} 7322 7323bool AudioFlinger::EffectModule::isProcessEnabled() const 7324{ 7325 switch (mState) { 7326 case RESTART: 7327 case ACTIVE: 7328 case STOPPING: 7329 case STOPPED: 7330 return true; 7331 case IDLE: 7332 case STARTING: 7333 case DESTROYED: 7334 default: 7335 return false; 7336 } 7337} 7338 7339status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7340{ 7341 Mutex::Autolock _l(mLock); 7342 status_t status = NO_ERROR; 7343 7344 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7345 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7346 if (isProcessEnabled() && 7347 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7348 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7349 status_t cmdStatus; 7350 uint32_t volume[2]; 7351 uint32_t *pVolume = NULL; 7352 uint32_t size = sizeof(volume); 7353 volume[0] = *left; 7354 volume[1] = *right; 7355 if (controller) { 7356 pVolume = volume; 7357 } 7358 status = (*mEffectInterface)->command(mEffectInterface, 7359 EFFECT_CMD_SET_VOLUME, 7360 size, 7361 volume, 7362 &size, 7363 pVolume); 7364 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7365 *left = volume[0]; 7366 *right = volume[1]; 7367 } 7368 } 7369 return status; 7370} 7371 7372status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7373{ 7374 Mutex::Autolock _l(mLock); 7375 status_t status = NO_ERROR; 7376 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7377 // audio pre processing modules on RecordThread can receive both output and 7378 // input device indication in the same call 7379 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7380 if (dev) { 7381 status_t cmdStatus; 7382 uint32_t size = sizeof(status_t); 7383 7384 status = (*mEffectInterface)->command(mEffectInterface, 7385 EFFECT_CMD_SET_DEVICE, 7386 sizeof(uint32_t), 7387 &dev, 7388 &size, 7389 &cmdStatus); 7390 if (status == NO_ERROR) { 7391 status = cmdStatus; 7392 } 7393 } 7394 dev = device & AUDIO_DEVICE_IN_ALL; 7395 if (dev) { 7396 status_t cmdStatus; 7397 uint32_t size = sizeof(status_t); 7398 7399 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7400 EFFECT_CMD_SET_INPUT_DEVICE, 7401 sizeof(uint32_t), 7402 &dev, 7403 &size, 7404 &cmdStatus); 7405 if (status2 == NO_ERROR) { 7406 status2 = cmdStatus; 7407 } 7408 if (status == NO_ERROR) { 7409 status = status2; 7410 } 7411 } 7412 } 7413 return status; 7414} 7415 7416status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7417{ 7418 Mutex::Autolock _l(mLock); 7419 status_t status = NO_ERROR; 7420 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7421 status_t cmdStatus; 7422 uint32_t size = sizeof(status_t); 7423 status = (*mEffectInterface)->command(mEffectInterface, 7424 EFFECT_CMD_SET_AUDIO_MODE, 7425 sizeof(audio_mode_t), 7426 &mode, 7427 &size, 7428 &cmdStatus); 7429 if (status == NO_ERROR) { 7430 status = cmdStatus; 7431 } 7432 } 7433 return status; 7434} 7435 7436void AudioFlinger::EffectModule::setSuspended(bool suspended) 7437{ 7438 Mutex::Autolock _l(mLock); 7439 mSuspended = suspended; 7440} 7441 7442bool AudioFlinger::EffectModule::suspended() const 7443{ 7444 Mutex::Autolock _l(mLock); 7445 return mSuspended; 7446} 7447 7448status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7449{ 7450 const size_t SIZE = 256; 7451 char buffer[SIZE]; 7452 String8 result; 7453 7454 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7455 result.append(buffer); 7456 7457 bool locked = tryLock(mLock); 7458 // failed to lock - AudioFlinger is probably deadlocked 7459 if (!locked) { 7460 result.append("\t\tCould not lock Fx mutex:\n"); 7461 } 7462 7463 result.append("\t\tSession Status State Engine:\n"); 7464 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7465 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7466 result.append(buffer); 7467 7468 result.append("\t\tDescriptor:\n"); 7469 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7470 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7471 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7472 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7473 result.append(buffer); 7474 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7475 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7476 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7477 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7478 result.append(buffer); 7479 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7480 mDescriptor.apiVersion, 7481 mDescriptor.flags); 7482 result.append(buffer); 7483 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7484 mDescriptor.name); 7485 result.append(buffer); 7486 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7487 mDescriptor.implementor); 7488 result.append(buffer); 7489 7490 result.append("\t\t- Input configuration:\n"); 7491 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7492 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7493 (uint32_t)mConfig.inputCfg.buffer.raw, 7494 mConfig.inputCfg.buffer.frameCount, 7495 mConfig.inputCfg.samplingRate, 7496 mConfig.inputCfg.channels, 7497 mConfig.inputCfg.format); 7498 result.append(buffer); 7499 7500 result.append("\t\t- Output configuration:\n"); 7501 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7502 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7503 (uint32_t)mConfig.outputCfg.buffer.raw, 7504 mConfig.outputCfg.buffer.frameCount, 7505 mConfig.outputCfg.samplingRate, 7506 mConfig.outputCfg.channels, 7507 mConfig.outputCfg.format); 7508 result.append(buffer); 7509 7510 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7511 result.append(buffer); 7512 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7513 for (size_t i = 0; i < mHandles.size(); ++i) { 7514 sp<EffectHandle> handle = mHandles[i].promote(); 7515 if (handle != 0) { 7516 handle->dump(buffer, SIZE); 7517 result.append(buffer); 7518 } 7519 } 7520 7521 result.append("\n"); 7522 7523 write(fd, result.string(), result.length()); 7524 7525 if (locked) { 7526 mLock.unlock(); 7527 } 7528 7529 return NO_ERROR; 7530} 7531 7532// ---------------------------------------------------------------------------- 7533// EffectHandle implementation 7534// ---------------------------------------------------------------------------- 7535 7536#undef LOG_TAG 7537#define LOG_TAG "AudioFlinger::EffectHandle" 7538 7539AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7540 const sp<AudioFlinger::Client>& client, 7541 const sp<IEffectClient>& effectClient, 7542 int32_t priority) 7543 : BnEffect(), 7544 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7545 mPriority(priority), mHasControl(false), mEnabled(false) 7546{ 7547 ALOGV("constructor %p", this); 7548 7549 if (client == 0) { 7550 return; 7551 } 7552 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7553 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7554 if (mCblkMemory != 0) { 7555 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7556 7557 if (mCblk != NULL) { 7558 new(mCblk) effect_param_cblk_t(); 7559 mBuffer = (uint8_t *)mCblk + bufOffset; 7560 } 7561 } else { 7562 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7563 return; 7564 } 7565} 7566 7567AudioFlinger::EffectHandle::~EffectHandle() 7568{ 7569 ALOGV("Destructor %p", this); 7570 disconnect(false); 7571 ALOGV("Destructor DONE %p", this); 7572} 7573 7574status_t AudioFlinger::EffectHandle::enable() 7575{ 7576 ALOGV("enable %p", this); 7577 if (!mHasControl) return INVALID_OPERATION; 7578 if (mEffect == 0) return DEAD_OBJECT; 7579 7580 if (mEnabled) { 7581 return NO_ERROR; 7582 } 7583 7584 mEnabled = true; 7585 7586 sp<ThreadBase> thread = mEffect->thread().promote(); 7587 if (thread != 0) { 7588 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7589 } 7590 7591 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7592 if (mEffect->suspended()) { 7593 return NO_ERROR; 7594 } 7595 7596 status_t status = mEffect->setEnabled(true); 7597 if (status != NO_ERROR) { 7598 if (thread != 0) { 7599 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7600 } 7601 mEnabled = false; 7602 } 7603 return status; 7604} 7605 7606status_t AudioFlinger::EffectHandle::disable() 7607{ 7608 ALOGV("disable %p", this); 7609 if (!mHasControl) return INVALID_OPERATION; 7610 if (mEffect == 0) return DEAD_OBJECT; 7611 7612 if (!mEnabled) { 7613 return NO_ERROR; 7614 } 7615 mEnabled = false; 7616 7617 if (mEffect->suspended()) { 7618 return NO_ERROR; 7619 } 7620 7621 status_t status = mEffect->setEnabled(false); 7622 7623 sp<ThreadBase> thread = mEffect->thread().promote(); 7624 if (thread != 0) { 7625 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7626 } 7627 7628 return status; 7629} 7630 7631void AudioFlinger::EffectHandle::disconnect() 7632{ 7633 disconnect(true); 7634} 7635 7636void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7637{ 7638 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7639 if (mEffect == 0) { 7640 return; 7641 } 7642 mEffect->disconnect(this, unpinIfLast); 7643 7644 if (mHasControl && mEnabled) { 7645 sp<ThreadBase> thread = mEffect->thread().promote(); 7646 if (thread != 0) { 7647 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7648 } 7649 } 7650 7651 // release sp on module => module destructor can be called now 7652 mEffect.clear(); 7653 if (mClient != 0) { 7654 if (mCblk != NULL) { 7655 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7656 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7657 } 7658 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7659 // Client destructor must run with AudioFlinger mutex locked 7660 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7661 mClient.clear(); 7662 } 7663} 7664 7665status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7666 uint32_t cmdSize, 7667 void *pCmdData, 7668 uint32_t *replySize, 7669 void *pReplyData) 7670{ 7671// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7672// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7673 7674 // only get parameter command is permitted for applications not controlling the effect 7675 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7676 return INVALID_OPERATION; 7677 } 7678 if (mEffect == 0) return DEAD_OBJECT; 7679 if (mClient == 0) return INVALID_OPERATION; 7680 7681 // handle commands that are not forwarded transparently to effect engine 7682 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7683 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7684 // no risk to block the whole media server process or mixer threads is we are stuck here 7685 Mutex::Autolock _l(mCblk->lock); 7686 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7687 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7688 mCblk->serverIndex = 0; 7689 mCblk->clientIndex = 0; 7690 return BAD_VALUE; 7691 } 7692 status_t status = NO_ERROR; 7693 while (mCblk->serverIndex < mCblk->clientIndex) { 7694 int reply; 7695 uint32_t rsize = sizeof(int); 7696 int *p = (int *)(mBuffer + mCblk->serverIndex); 7697 int size = *p++; 7698 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7699 ALOGW("command(): invalid parameter block size"); 7700 break; 7701 } 7702 effect_param_t *param = (effect_param_t *)p; 7703 if (param->psize == 0 || param->vsize == 0) { 7704 ALOGW("command(): null parameter or value size"); 7705 mCblk->serverIndex += size; 7706 continue; 7707 } 7708 uint32_t psize = sizeof(effect_param_t) + 7709 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7710 param->vsize; 7711 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7712 psize, 7713 p, 7714 &rsize, 7715 &reply); 7716 // stop at first error encountered 7717 if (ret != NO_ERROR) { 7718 status = ret; 7719 *(int *)pReplyData = reply; 7720 break; 7721 } else if (reply != NO_ERROR) { 7722 *(int *)pReplyData = reply; 7723 break; 7724 } 7725 mCblk->serverIndex += size; 7726 } 7727 mCblk->serverIndex = 0; 7728 mCblk->clientIndex = 0; 7729 return status; 7730 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7731 *(int *)pReplyData = NO_ERROR; 7732 return enable(); 7733 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7734 *(int *)pReplyData = NO_ERROR; 7735 return disable(); 7736 } 7737 7738 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7739} 7740 7741void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7742{ 7743 ALOGV("setControl %p control %d", this, hasControl); 7744 7745 mHasControl = hasControl; 7746 mEnabled = enabled; 7747 7748 if (signal && mEffectClient != 0) { 7749 mEffectClient->controlStatusChanged(hasControl); 7750 } 7751} 7752 7753void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7754 uint32_t cmdSize, 7755 void *pCmdData, 7756 uint32_t replySize, 7757 void *pReplyData) 7758{ 7759 if (mEffectClient != 0) { 7760 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7761 } 7762} 7763 7764 7765 7766void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7767{ 7768 if (mEffectClient != 0) { 7769 mEffectClient->enableStatusChanged(enabled); 7770 } 7771} 7772 7773status_t AudioFlinger::EffectHandle::onTransact( 7774 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7775{ 7776 return BnEffect::onTransact(code, data, reply, flags); 7777} 7778 7779 7780void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7781{ 7782 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7783 7784 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7785 (mClient == 0) ? getpid_cached : mClient->pid(), 7786 mPriority, 7787 mHasControl, 7788 !locked, 7789 mCblk ? mCblk->clientIndex : 0, 7790 mCblk ? mCblk->serverIndex : 0 7791 ); 7792 7793 if (locked) { 7794 mCblk->lock.unlock(); 7795 } 7796} 7797 7798#undef LOG_TAG 7799#define LOG_TAG "AudioFlinger::EffectChain" 7800 7801AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7802 int sessionId) 7803 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7804 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7805 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7806{ 7807 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7808 if (thread == NULL) { 7809 return; 7810 } 7811 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7812 thread->frameCount(); 7813} 7814 7815AudioFlinger::EffectChain::~EffectChain() 7816{ 7817 if (mOwnInBuffer) { 7818 delete mInBuffer; 7819 } 7820 7821} 7822 7823// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7824sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7825{ 7826 size_t size = mEffects.size(); 7827 7828 for (size_t i = 0; i < size; i++) { 7829 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7830 return mEffects[i]; 7831 } 7832 } 7833 return 0; 7834} 7835 7836// getEffectFromId_l() must be called with ThreadBase::mLock held 7837sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7838{ 7839 size_t size = mEffects.size(); 7840 7841 for (size_t i = 0; i < size; i++) { 7842 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7843 if (id == 0 || mEffects[i]->id() == id) { 7844 return mEffects[i]; 7845 } 7846 } 7847 return 0; 7848} 7849 7850// getEffectFromType_l() must be called with ThreadBase::mLock held 7851sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7852 const effect_uuid_t *type) 7853{ 7854 size_t size = mEffects.size(); 7855 7856 for (size_t i = 0; i < size; i++) { 7857 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7858 return mEffects[i]; 7859 } 7860 } 7861 return 0; 7862} 7863 7864// Must be called with EffectChain::mLock locked 7865void AudioFlinger::EffectChain::process_l() 7866{ 7867 sp<ThreadBase> thread = mThread.promote(); 7868 if (thread == 0) { 7869 ALOGW("process_l(): cannot promote mixer thread"); 7870 return; 7871 } 7872 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7873 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7874 // always process effects unless no more tracks are on the session and the effect tail 7875 // has been rendered 7876 bool doProcess = true; 7877 if (!isGlobalSession) { 7878 bool tracksOnSession = (trackCnt() != 0); 7879 7880 if (!tracksOnSession && mTailBufferCount == 0) { 7881 doProcess = false; 7882 } 7883 7884 if (activeTrackCnt() == 0) { 7885 // if no track is active and the effect tail has not been rendered, 7886 // the input buffer must be cleared here as the mixer process will not do it 7887 if (tracksOnSession || mTailBufferCount > 0) { 7888 size_t numSamples = thread->frameCount() * thread->channelCount(); 7889 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7890 if (mTailBufferCount > 0) { 7891 mTailBufferCount--; 7892 } 7893 } 7894 } 7895 } 7896 7897 size_t size = mEffects.size(); 7898 if (doProcess) { 7899 for (size_t i = 0; i < size; i++) { 7900 mEffects[i]->process(); 7901 } 7902 } 7903 for (size_t i = 0; i < size; i++) { 7904 mEffects[i]->updateState(); 7905 } 7906} 7907 7908// addEffect_l() must be called with PlaybackThread::mLock held 7909status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7910{ 7911 effect_descriptor_t desc = effect->desc(); 7912 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7913 7914 Mutex::Autolock _l(mLock); 7915 effect->setChain(this); 7916 sp<ThreadBase> thread = mThread.promote(); 7917 if (thread == 0) { 7918 return NO_INIT; 7919 } 7920 effect->setThread(thread); 7921 7922 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7923 // Auxiliary effects are inserted at the beginning of mEffects vector as 7924 // they are processed first and accumulated in chain input buffer 7925 mEffects.insertAt(effect, 0); 7926 7927 // the input buffer for auxiliary effect contains mono samples in 7928 // 32 bit format. This is to avoid saturation in AudoMixer 7929 // accumulation stage. Saturation is done in EffectModule::process() before 7930 // calling the process in effect engine 7931 size_t numSamples = thread->frameCount(); 7932 int32_t *buffer = new int32_t[numSamples]; 7933 memset(buffer, 0, numSamples * sizeof(int32_t)); 7934 effect->setInBuffer((int16_t *)buffer); 7935 // auxiliary effects output samples to chain input buffer for further processing 7936 // by insert effects 7937 effect->setOutBuffer(mInBuffer); 7938 } else { 7939 // Insert effects are inserted at the end of mEffects vector as they are processed 7940 // after track and auxiliary effects. 7941 // Insert effect order as a function of indicated preference: 7942 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7943 // another effect is present 7944 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7945 // last effect claiming first position 7946 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7947 // first effect claiming last position 7948 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7949 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7950 // already present 7951 7952 size_t size = mEffects.size(); 7953 size_t idx_insert = size; 7954 ssize_t idx_insert_first = -1; 7955 ssize_t idx_insert_last = -1; 7956 7957 for (size_t i = 0; i < size; i++) { 7958 effect_descriptor_t d = mEffects[i]->desc(); 7959 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7960 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7961 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7962 // check invalid effect chaining combinations 7963 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7964 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7965 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7966 return INVALID_OPERATION; 7967 } 7968 // remember position of first insert effect and by default 7969 // select this as insert position for new effect 7970 if (idx_insert == size) { 7971 idx_insert = i; 7972 } 7973 // remember position of last insert effect claiming 7974 // first position 7975 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7976 idx_insert_first = i; 7977 } 7978 // remember position of first insert effect claiming 7979 // last position 7980 if (iPref == EFFECT_FLAG_INSERT_LAST && 7981 idx_insert_last == -1) { 7982 idx_insert_last = i; 7983 } 7984 } 7985 } 7986 7987 // modify idx_insert from first position if needed 7988 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7989 if (idx_insert_last != -1) { 7990 idx_insert = idx_insert_last; 7991 } else { 7992 idx_insert = size; 7993 } 7994 } else { 7995 if (idx_insert_first != -1) { 7996 idx_insert = idx_insert_first + 1; 7997 } 7998 } 7999 8000 // always read samples from chain input buffer 8001 effect->setInBuffer(mInBuffer); 8002 8003 // if last effect in the chain, output samples to chain 8004 // output buffer, otherwise to chain input buffer 8005 if (idx_insert == size) { 8006 if (idx_insert != 0) { 8007 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8008 mEffects[idx_insert-1]->configure(); 8009 } 8010 effect->setOutBuffer(mOutBuffer); 8011 } else { 8012 effect->setOutBuffer(mInBuffer); 8013 } 8014 mEffects.insertAt(effect, idx_insert); 8015 8016 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8017 } 8018 effect->configure(); 8019 return NO_ERROR; 8020} 8021 8022// removeEffect_l() must be called with PlaybackThread::mLock held 8023size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8024{ 8025 Mutex::Autolock _l(mLock); 8026 size_t size = mEffects.size(); 8027 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8028 8029 for (size_t i = 0; i < size; i++) { 8030 if (effect == mEffects[i]) { 8031 // calling stop here will remove pre-processing effect from the audio HAL. 8032 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8033 // the middle of a read from audio HAL 8034 if (mEffects[i]->state() == EffectModule::ACTIVE || 8035 mEffects[i]->state() == EffectModule::STOPPING) { 8036 mEffects[i]->stop(); 8037 } 8038 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8039 delete[] effect->inBuffer(); 8040 } else { 8041 if (i == size - 1 && i != 0) { 8042 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8043 mEffects[i - 1]->configure(); 8044 } 8045 } 8046 mEffects.removeAt(i); 8047 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8048 break; 8049 } 8050 } 8051 8052 return mEffects.size(); 8053} 8054 8055// setDevice_l() must be called with PlaybackThread::mLock held 8056void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8057{ 8058 size_t size = mEffects.size(); 8059 for (size_t i = 0; i < size; i++) { 8060 mEffects[i]->setDevice(device); 8061 } 8062} 8063 8064// setMode_l() must be called with PlaybackThread::mLock held 8065void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8066{ 8067 size_t size = mEffects.size(); 8068 for (size_t i = 0; i < size; i++) { 8069 mEffects[i]->setMode(mode); 8070 } 8071} 8072 8073// setVolume_l() must be called with PlaybackThread::mLock held 8074bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8075{ 8076 uint32_t newLeft = *left; 8077 uint32_t newRight = *right; 8078 bool hasControl = false; 8079 int ctrlIdx = -1; 8080 size_t size = mEffects.size(); 8081 8082 // first update volume controller 8083 for (size_t i = size; i > 0; i--) { 8084 if (mEffects[i - 1]->isProcessEnabled() && 8085 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8086 ctrlIdx = i - 1; 8087 hasControl = true; 8088 break; 8089 } 8090 } 8091 8092 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8093 if (hasControl) { 8094 *left = mNewLeftVolume; 8095 *right = mNewRightVolume; 8096 } 8097 return hasControl; 8098 } 8099 8100 mVolumeCtrlIdx = ctrlIdx; 8101 mLeftVolume = newLeft; 8102 mRightVolume = newRight; 8103 8104 // second get volume update from volume controller 8105 if (ctrlIdx >= 0) { 8106 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8107 mNewLeftVolume = newLeft; 8108 mNewRightVolume = newRight; 8109 } 8110 // then indicate volume to all other effects in chain. 8111 // Pass altered volume to effects before volume controller 8112 // and requested volume to effects after controller 8113 uint32_t lVol = newLeft; 8114 uint32_t rVol = newRight; 8115 8116 for (size_t i = 0; i < size; i++) { 8117 if ((int)i == ctrlIdx) continue; 8118 // this also works for ctrlIdx == -1 when there is no volume controller 8119 if ((int)i > ctrlIdx) { 8120 lVol = *left; 8121 rVol = *right; 8122 } 8123 mEffects[i]->setVolume(&lVol, &rVol, false); 8124 } 8125 *left = newLeft; 8126 *right = newRight; 8127 8128 return hasControl; 8129} 8130 8131status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8132{ 8133 const size_t SIZE = 256; 8134 char buffer[SIZE]; 8135 String8 result; 8136 8137 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8138 result.append(buffer); 8139 8140 bool locked = tryLock(mLock); 8141 // failed to lock - AudioFlinger is probably deadlocked 8142 if (!locked) { 8143 result.append("\tCould not lock mutex:\n"); 8144 } 8145 8146 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8147 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8148 mEffects.size(), 8149 (uint32_t)mInBuffer, 8150 (uint32_t)mOutBuffer, 8151 mActiveTrackCnt); 8152 result.append(buffer); 8153 write(fd, result.string(), result.size()); 8154 8155 for (size_t i = 0; i < mEffects.size(); ++i) { 8156 sp<EffectModule> effect = mEffects[i]; 8157 if (effect != 0) { 8158 effect->dump(fd, args); 8159 } 8160 } 8161 8162 if (locked) { 8163 mLock.unlock(); 8164 } 8165 8166 return NO_ERROR; 8167} 8168 8169// must be called with ThreadBase::mLock held 8170void AudioFlinger::EffectChain::setEffectSuspended_l( 8171 const effect_uuid_t *type, bool suspend) 8172{ 8173 sp<SuspendedEffectDesc> desc; 8174 // use effect type UUID timelow as key as there is no real risk of identical 8175 // timeLow fields among effect type UUIDs. 8176 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8177 if (suspend) { 8178 if (index >= 0) { 8179 desc = mSuspendedEffects.valueAt(index); 8180 } else { 8181 desc = new SuspendedEffectDesc(); 8182 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8183 mSuspendedEffects.add(type->timeLow, desc); 8184 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8185 } 8186 if (desc->mRefCount++ == 0) { 8187 sp<EffectModule> effect = getEffectIfEnabled(type); 8188 if (effect != 0) { 8189 desc->mEffect = effect; 8190 effect->setSuspended(true); 8191 effect->setEnabled(false); 8192 } 8193 } 8194 } else { 8195 if (index < 0) { 8196 return; 8197 } 8198 desc = mSuspendedEffects.valueAt(index); 8199 if (desc->mRefCount <= 0) { 8200 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8201 desc->mRefCount = 1; 8202 } 8203 if (--desc->mRefCount == 0) { 8204 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8205 if (desc->mEffect != 0) { 8206 sp<EffectModule> effect = desc->mEffect.promote(); 8207 if (effect != 0) { 8208 effect->setSuspended(false); 8209 sp<EffectHandle> handle = effect->controlHandle(); 8210 if (handle != 0) { 8211 effect->setEnabled(handle->enabled()); 8212 } 8213 } 8214 desc->mEffect.clear(); 8215 } 8216 mSuspendedEffects.removeItemsAt(index); 8217 } 8218 } 8219} 8220 8221// must be called with ThreadBase::mLock held 8222void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8223{ 8224 sp<SuspendedEffectDesc> desc; 8225 8226 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8227 if (suspend) { 8228 if (index >= 0) { 8229 desc = mSuspendedEffects.valueAt(index); 8230 } else { 8231 desc = new SuspendedEffectDesc(); 8232 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8233 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8234 } 8235 if (desc->mRefCount++ == 0) { 8236 Vector< sp<EffectModule> > effects; 8237 getSuspendEligibleEffects(effects); 8238 for (size_t i = 0; i < effects.size(); i++) { 8239 setEffectSuspended_l(&effects[i]->desc().type, true); 8240 } 8241 } 8242 } else { 8243 if (index < 0) { 8244 return; 8245 } 8246 desc = mSuspendedEffects.valueAt(index); 8247 if (desc->mRefCount <= 0) { 8248 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8249 desc->mRefCount = 1; 8250 } 8251 if (--desc->mRefCount == 0) { 8252 Vector<const effect_uuid_t *> types; 8253 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8254 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8255 continue; 8256 } 8257 types.add(&mSuspendedEffects.valueAt(i)->mType); 8258 } 8259 for (size_t i = 0; i < types.size(); i++) { 8260 setEffectSuspended_l(types[i], false); 8261 } 8262 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8263 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8264 } 8265 } 8266} 8267 8268 8269// The volume effect is used for automated tests only 8270#ifndef OPENSL_ES_H_ 8271static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8272 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8273const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8274#endif //OPENSL_ES_H_ 8275 8276bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8277{ 8278 // auxiliary effects and visualizer are never suspended on output mix 8279 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8280 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8281 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8282 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8283 return false; 8284 } 8285 return true; 8286} 8287 8288void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8289{ 8290 effects.clear(); 8291 for (size_t i = 0; i < mEffects.size(); i++) { 8292 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8293 effects.add(mEffects[i]); 8294 } 8295 } 8296} 8297 8298sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8299 const effect_uuid_t *type) 8300{ 8301 sp<EffectModule> effect = getEffectFromType_l(type); 8302 return effect != 0 && effect->isEnabled() ? effect : 0; 8303} 8304 8305void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8306 bool enabled) 8307{ 8308 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8309 if (enabled) { 8310 if (index < 0) { 8311 // if the effect is not suspend check if all effects are suspended 8312 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8313 if (index < 0) { 8314 return; 8315 } 8316 if (!isEffectEligibleForSuspend(effect->desc())) { 8317 return; 8318 } 8319 setEffectSuspended_l(&effect->desc().type, enabled); 8320 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8321 if (index < 0) { 8322 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8323 return; 8324 } 8325 } 8326 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8327 effect->desc().type.timeLow); 8328 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8329 // if effect is requested to suspended but was not yet enabled, supend it now. 8330 if (desc->mEffect == 0) { 8331 desc->mEffect = effect; 8332 effect->setEnabled(false); 8333 effect->setSuspended(true); 8334 } 8335 } else { 8336 if (index < 0) { 8337 return; 8338 } 8339 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8340 effect->desc().type.timeLow); 8341 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8342 desc->mEffect.clear(); 8343 effect->setSuspended(false); 8344 } 8345} 8346 8347#undef LOG_TAG 8348#define LOG_TAG "AudioFlinger" 8349 8350// ---------------------------------------------------------------------------- 8351 8352status_t AudioFlinger::onTransact( 8353 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8354{ 8355 return BnAudioFlinger::onTransact(code, data, reply, flags); 8356} 8357 8358}; // namespace android 8359